This chapter covers session configuration objects, N through Z, alphabetically. For session configuration objects A through M, see "Configuring Session Configuration Objects A Through M." The session configuration objects define the way in which the ME handles SIP-based signaling and media traffic. The session configuration that is applied to an active call through the ME depends on configuration of other aspects of the system.
There are several places in the configuration hierarchy through which you can access the session configuration objects. The path to these object defines in which cases the ME uses that configuration. Locations for session configuration are defined in the following table.
Table 40-1 Session Configuration Locations
| Path | Defines... |
|---|---|
|
vsp > default-session-config |
The session configuration settings to apply to those SIP calls for which there are no configured policies. See Configuring Default Session Configuration Objects, for more information. |
|
vsp > policies > session-policies > policy > rule |
The session configuration settings to apply to SIP calls for which a configured policy exists. See Configuring Policy Objects, for more information. |
|
vsp > dial-plan > dial-prefix vsp > dial-plan > route vsp > dial-plan > source-route |
The session configuration settings to apply to calls based on the dial prefix or domain suffix. See Configuring Dial Plan Objects, for more information. |
|
vsp > calling-group > route vsp > calling-group > source-route |
The session configuration settings to apply to calling-group member calls based on the dial prefix or domain suffix. See Configuring Calling Group Objects, for more information. |
|
vsp > session-config-pool > entry |
A saved session configuration that can be referenced within one or more dial plans. See Configuring Session Configuration Pool Objects, for more information. |
The ME supports a generic database used to hold named variables. A named variable is a variable paired with a value through the reg-exp header code. This allows you to modify SIP message fields and CDR fields more generically.
This object allows you to configure the ME to look at any SIP message header, and by applying regular expressions, extract any part of the header and store the value in the specified named variable table. You can apply these settings to both requests and responses.
Under this object you assign the named variable collector a number and a name. This is also where you create the variable. To create a variable, first select the header type you want to serve as the source of the data. This can be either any valid SIP message header. Then specify a regular expression to run against the value of the source header. Finally, specify the replacement expression to apply when there is a match. If you want to append a string to the existing named-variable value, appending is done the same way as creating.
Other things you can configure under the named-variable-collector object include the SIP methods to apply variables, whether the ME applies variables to responses, which type of dialog variables should be applied, if a CSEQ field needs to be matched, and what actions to take on the expression if there is not a complete match.
When a session is created, a named variable list is automatically created. If any named variables are configured in the default-session-config, they populate this list. All variable names in the named variable list must be unique.
The ME updates the named variable list when any of the following happens.
When the session configuration merge-object is set to merge, the named variables configured in the new session config are appended to the existing named variable list.
When the session configuration merge-named-variables is set to replace, the existing session configuration named variables are replaced by the newly configured named variables.
When the header-settings > named-variable-collector collects new named variables via the reg-exp code.
When you specify or create a named variable list and a named variable of the same name already exists, the ME overwrites the value of the existing variable name.
Note:
Variable names cannot start with a ”$” and you should not use special characters such as ”\”, ”%”, ”#”, ”!”, ”?”, ”[”, ”]”, :&”, ”{”, ”}”, ”@” when naming variables.admin: Enables or disables named variable collection on the ME.
Default: enabled
Values: enabled | disabled
Example: set admin enabled
number: Specifies an indexed number for this entry.
Default: There is no default setting
Values: Min: / Max: 4294967296
Example: set number 10
named-variable: Specifies the named-variable you are creating or modifying. The value of this variable will be the replacement string.
Default: There is no default setting
Example: set named-variable var1
create: Identifies the header containing the data to be modified and the value assigned to the named-variable. First, select the header that serves as the source of the data. This can be To, From, or Request. Then specify a regular expression to run against the value of the source header. Finally, supply the replacement expression to apply if there is a match.
Default: There is no default setting
Example: set create To (.*) \1;\1>
append: Identifies the header containing the data to be added to the named-variable value. First, select the header that serves as the source of the data. This can be To, From, or Request. Then specify a regular expression to run against the value of the source header. Finally, supply the replacement expression to apply if there is a match. The ME appends this string to the existing named-variable value. To add spaces or commas, include them using quotation marks.
Default: There is no default setting
Example: set append From (.*) \2;\2>
apply-to-methods: Specifies the message type from which the ME extracts the specified named-variable value.
Default: INVITE
Example: set apply-to-methods REGISTER
apply-to-responses: Specifies whether to apply header value changes to SIP requests or both requests and responses. When set to no, changes are only applied to requests. When set to yes, changes are applied to both requests and responses.
Default: no
Values: no
yes [response-code]
both [response-code]
Example: set apply-to-responses both 500
apply-to-dialog: Specifies whether to apply header value changes to a specific dialog or not.
Default: both
Values: inbound: Apply to the inbound dialog only
outbound: Apply to the outbound dialog only
both: Apply to both inbound and outbound dialogs.
Example: set apply-to-dialog inbound
cseq: Secondary property. Sets a mechanism to further filter which SIP messages have the header expression modifications applied. If this property is set to 0 (the default), the ME applies the changes to all SIP messages. If set to any other value, the ME only applies the changes to SIP messages having a CSEQ field that matches that value.
Default: 0
Values: Min: 0 / Max: 4294967296
Example: set cseq 100
create-on-failed-match: Secondary property. Construct a create header even when the expression is not a complete match.
Default: true
Values: true | false
Example: set create-on-failed-match false
append-on-failed-match: Secondary property. Execute the append action event even when the create expression fails to match.
Default: true
Values: true | false
Example: set append-on-failed-match false
The named-variables object, configured on a per-session basis, creates a static list of named variables that can be used in several features throughout the ME configuration. You create named variables under the session-config object and you can create a named variable list for each configured session.
config vsp session-config-pool entry named-variables config vsp default-session-config named-variables
named-variable: Enter the name of the named variable for this session.
Default: There is no default setting
Example: set
value: Enter the value of the named variable for this session.
Default: There is no default setting
Example: set value 100
merge-object: Specifies whether this new set of session configuration named variables overwrites the previous set of session configuration named variables, or if it is appended to the previous set.
Default: merge
Values: merge: Append to the existing set of session configuration named variables.
replace: Overwrite the existing set of session configuration named variables with the new one.
Example: set merge-object replace
Specifies whether symmetric Real-time Transport Protocol (RTP) is applied to the session. When a SIP call passes through far-end NAT to reach the call recipient, the complications of NAT (or a firewall) create problems for call connections.To address this, you can configure the ME to run symmetric RTP.
RTP is a packet-based communication protocol that adds timing and sequence information to provide end-to-end network transport functions for applications transmitting real-time data, such as audio or video. With symmetric RTP on, the ME sends return RTP messages based on the source IP address and UDP port in received RTP messages. NAT only modifies data in the IP header: the Session Description Protocol (SDP) payload is left unchanged. By using the source IP address and UDP port from the received RTP message, the ME sends traffic back to the NAT device, instead of the untranslated addresses in the SDP message.
config vsp default-session-config media nat-traversal config vsp policies session-policies policy name rule name session-config media nat-traversal config vsp dial-plan dial-prefix entryName session-config media nat-traversal config vsp dial-plan route name session-config media nat-traversal config vsp dial-plan source-route name session-config media nat-traversal config vsp session-config-pool entry name media nat-traversal
admin: Enables or disables the NAT traversal configuration on this ME device. When enabled, the system uses the settings configured here; when disabled the system uses the default NAT traversal settings.
Default: enabled
Values: enabled | disabled
Example: set admin disabled
symmetricRTP: Specifies whether to use symmetric RTP for SIP call sessions. It specifies how the system learns the public address in preference to the private address (behind the firewall). When true, the system uses the address and port numbers that it learns from received RTP packets as the destination address for the endpoint. When set to false, the system uses the address and port numbers that it learns from the SDP message as the destination address for the endpoint.
Default: false
Values: true | false
Example: set symmetricRTP true
asymmetric-rtp-address: Specifies addresses that are not compatible with symmetric RTP. When an address that matches this property is received in an SDP message, the system disables symmetric RTP when sending RTP to that endpoint. This property is only necessary in certain situations (e.g., when using the BroadWorks Media Server), and only when the symmetricRTP property is set to true.
Default: There is no default setting
Example: set asymmetric-rt-address 192.168.10.10/32
Sets a preference for CODECs, influencing the ME ordering of CODECs in the SDC on the outbound leg of a call. The ME removes those CODECs with a zero priority from the SDP. CODEC preferences do not cause the ME to add a CODEC to the SDP, but to remove and/or reorder existing CODECs according to their priority. The ME places a CODEC whose priority is not specified in its original order, just ahead of the known auxiliary CODECs (e.g., telephone-events). This is not a mechanism to add CODECs into an SDP, only to order those that are already there (via transcoding or from the original offer/answer).
config vsp default-session-config out-codec-preferences config vsp policies session-policies policy name rule name session-config out-codec-preferences config vsp dial-plan dial-prefix entryName session-config out-codec-preferences config vsp dial-plan route name session-config out-codec-preferences config vsp dial-plan source-route name session-config out-codec-preferences config vsp session-config-pool entry name out-codec-preferences
preferences<media-type>[codec][priority]: Assigns a priority to a given CODEC for outbound audio or video sessions.
Default: There are no default settings
Values: media type: Select for audio or video. The associated CODEC (subtype) is preferred according to the priority for that media type.
codec: The CODEC to which the priority applies. Use the question mark character at the command line to see a list of available CODECs, or enter any CODEC name.
priority: Sets a preference for the CODEC. The lower the number, the more preferred the CODEC. Assigning a priority value of zero disables the CODEC for the session. The system removes these CODECs before sending the SDP offer or answer. (0-100)
Example: set preferences video g729 1
Sets the parameters for outbound encrypted media sessions when the ME is anchoring the call. This is the portion from the ME to the call recipient. With this object you set the encryption requirements on the call, and the encryption method used. Because the ME does not always know on the outbound leg the encryption method expected by the recipient (because that recipient is not in the registry), you must manually set the type of encryption to offer.
When using RFC 1889, Microsoft clients do not typically do encryption unless it is offered as mandatory in the SDP by one of the clients. If you want encryption on the outbound side, you must set the mode property to require.
config vsp default-session-config out-encryption config vsp policies session-policies policy name rule name session-config out-encryption config vsp dial-plan dial-prefix entryName session-config out-encryption config vsp dial-plan route name session-config out-encryption config vsp dial-plan source-route name session-config out-encryption config vsp session-config-pool entry name out-encryption
mode: Specifies the encryption requirements on the outgoing call (from system to endpoint). The method of encryption used is determined by the type property.
Default: none
Values: none: The system disables the encryption put forth by the outbound endpoint (i.e., it responds ”no” to the encryption portion of the authentication handshake.) If the inbound endpoint requires encryption, the call is dropped.
pass-thru: The system passes cryptographic parameters through the box and does not participate in RTP encryption. This method renders some advanced media services unusable (in particular, recording, announcements, transcoding, call monitoring, RTP stats, media verification, and RTCP generation). When using this option, set mode to pass-thru in both in-encryption and out-encryption.
allow: If the outgoing endpoint offers encryption to the ME device, the system answers with it. If the endpoint does not offer encryption, the system does not answer with or initiate encryption.
follow: If the inbound endpoint offers encryption, the system offers encryption (the type set by policy) to the outbound endpoint.
offer: The system offers encryption to the outbound endpoint when the session is first established.
reoffer: The system offers encryption to the outbound endpoint whether the message is an INVITE or a REINVITE. This setting is most applicable when an endpoint issues a REINVITE, and encryption was not required with the original INVITE. In this case, the system will again offer encryption when forwarding the message.
require: The call must come in with encryption specified or the system drops it.
Example: set mode allow
type: Sets the type of encryption on outbound sessions. In choosing a type, the system uses the encryption expected by that device or application.
Default: RFC-3711
Values: Linksys: Uses SIP INFO messages to exchange mini-certificates and exchange a symmetric key. The media encryption is similar to RFC-3711, but done with AES-128 Countermode and with HMAC MD5 authentication. See the linksys action for more certificate information.
RFC-1889: Use encryption as defined in RFC 1889, RTP: A Transport Protocol for Real-Time Applications. This mode is used for compatibility with Windows Messenger and Microsoft Office Communicator, neither of which currently support RFC-3711 encryption. Instead, it uses a DES-CBC encryption of the entire UDP payload (including RTP headers) with no authentication. Note that to enable outbound encryption when using FRC 1889, you must set mode to require.
RFC-3711: Use encryption as defined in RFC 3711, The Secure Real-time Transport Protocol (SRTP).
Example: set type RFC-1899
require-tls: Specifies the requirements of the signaling protocol for a call outbound leg. It defines whether the system offers SRTP over a non-secure (TCP or UDP) signaling connection. The action of this property depends on the setting of the mode property.
Most phones follow RFC 4568, SDP Security Descriptions for Media Streams, and thus require that this property be set to true.
Default: false
Values: true: The system only offers encryption when talking to a TLS client. If TLS and SRTP are required (mode is set to require), the system fails calls going to TCP/UDP clients. If the mode property is set to offer or follow, the system forwards the call without SRTP.
false: The system offers SDP messages according to the mode setting without regard for the signaling transport. This allows keys to be exchanged in an insecure message.
Example: set require-tls true
priority-AES-128-CM-HMAC-SHA1-32: Secondary property. Sets a preference for 32-bit SHA1 authentication tags on outcoming calls. The system supports (offers) both 32- and 80-bit authentication tags on ingress. A value of 0 disables support for the 32-bit tag.
Default: 1
Values: Min: 1 (most preferred) / Max: 5
Example: set priority-AES-128-CM-HMAC-SHA1-32 2
priority-AES-128-CM-HMAC-SHA1-80: Secondary property. Sets a preference for 80-bit SHA1 authentication tags on outcoming calls. The system supports (offers) both 32- and 80-bit authentication tags on ingress. A value of 0 disables support for the 80-bit tag. To enable, set a value between 1 (most preferred) and 5.
Default: 2
Values: Min: 1 (most preferred) / Max: 5
Example: set priority-AES-128-CM-HMAC-SHA1-80 1
mki-length: Secondary property. Provides support for the optional Master Key Identifier bit defined in RFC 3711, The Secure Real-time Transport Protocol (SRTP). The value specify sets the number of bytes in the MKI. The system then sends the negotiated identifier of that length indicating which master key to use for decryption with each SRTP packet. Note that the endpoint must support this option.
Default: 0
Values: Min: 0 / Max: 4
Example: set mki-length 2
mikey-offer-location: Secondary property. Controls where in the SDP the system stores the MIKEY offer (the ”a=key-mgmt:mikey” line) when it is made. If MIKEY is offered to the system, it puts the MIKEY answer in the location where the offer was located. Set the location as the media descriptor or session level.
Default: session
Values: session | media-descriptor
Example: set mikey-offer-location media-descriptor
mikey-time-tolerance: Secondary property. Controls where in the SDP the system stores the MIKEY offer (the ”a=key-mgmt:mikey” line) when it is made. If MIKEY is offered to the system, it puts the MIKEY answer in the location where the offer was located. Set the location as the media descriptor or session level.
Default: 60
Example: set mikey-time-tolerance 90
symmetric-address-failure: Secondary property. Specifies whether the system learns the source IP address from the RTP/RTCP packets, even if the packets fail decryption. When enabled, the first packet in a particular stream that fails SRTP decryption causes a DroppedPacket notification to be sent to the application with the address of the packet. The application treats this like a srcIPChanged notification.
Default: disabled
Values: enabled | disabled
Example: set symmetric-address-failure enabled
treat-as-secure: Secondary property. Specifies whether a proprietary security indicator is used on the SIP interface; either ST-secure or ST-insecure. This setting only operates on the X-Siemends-Call-Type header when MIKEY encryption is involved in pass-thru mode.
Default: disabled
Values: disabled: Sets ST-insecure
enabled: Sets ST-secure
auto: If SRTP is active on both sides of the call, the ME allows the X-Siemends-Call-Type header to pass unchanged. If SRTP is not active on other side of the call, the header is set to ST-insecure. The ”auto” value sets the interface to trusted.
Example: set treat-as-secure enabled
Configures the ME's out-leg DTMF method preferences.
config vsp default-session-config out-dtmf-preferences config vsp session-config-pool entry out-dtmf-preferences
admin: Specifies whether or not this DTMF preference list is applied to calls matching this session configuration.
Default: disabled
Values: enabled | disabled
Example: set admin enabled
preferences: Allows you to configure supported dtmf-types and assign them with a priority to determine the ME's preferences.
Default: audio 1
First select a DTMF method. The available DTMF methods are:
Values: audio
rfc-2833
sip-info-dtmf
sip-info-dtmf-relay
sip-notify
h245-alphanumeric
h245-signal
q931
Then assign it a priority. This can be from 0-100. A value of 0 means the method is not supported. The lower the priority, the more preferred the DTMF method.
Example: set preferences rfc2833 2
The ME can be configured to translate one DTMF method to another. Through this object, you can control the length of play and pause time and volume for the digits that the ME plays on the out-leg.
config vsp default-session-config in-dtmf-settings config vsp session-config-pool entry out-dtmf-settings
digit-volume: Specifies the volume setting for the DTMF tones. The digit volume is measured in decibel (dB) of the measured power referenced to one milliwatt, measured at a zero transmission level point. The smaller the dBm0, the louder the volume.
Default: -20
Values: Min: -36 / Max: 0
digit-duration: Specifies the length of time, in milliseconds, that the ME plays each DTMF digit.
Default: 750
Values: Min: 100 / Max: 10000
Example: set digit-duration 1000
min-digit-duration: Specifies the minimum length of time, in milliseconds, that the ME plays each DTMF digit. If a DTMF event has a duration less than this value, the digit-duration property overrides the duration and is used to play the DTMF event.
Default: 60
Values: Min: 5 / Max: 100
Example: set min-digit-duration 75
max-digit-duration: Specifies the maximum length of time, in milliseconds, that the ME plays each DTMF digit. If a DTMF event has a duration greater than this value, the digit-duration property overrides the duration and is used to play the DTMF event.
Default: 2000
Values: Min: 100 / Max: 10000
Example: set max-digit-duration 3000
inter-digit-duration: Specifies the length of time, in milliseconds, that the ME pauses between playing each digit.
Default: 250
Values: Min: 0 / Max: 1000
Example: set inter-digit-duration 500
pause-duration: Specifies the length of time, in milliseconds that the ME pauses when it encounters a comma character in the conference code. The comma is a special character in the conference code that indicates a specified time the ME must wait before playing the next tone.
Default: 3000
Values: Min: 500 / Max: 10000
Example: set pause-duration 4000
minimum-duration: Specifies the minimum time, in milliseconds, between detecting RFC-2833 events.
Default: 60
Values: Min: 0 / Max: 1000
Example: set minimum-duration 100
as-audio: Specifies whether the ME sends audio or DTMF packets to the conference server when representing conference code tones. When true, the ME encodes the sound in the current CODEC. When false, the ME sends DTMF packets.
Default: true
Values: true | false
Example: set as-audio false
Secondary object. Controls the method used for forwarding DTMF tones in a call. The two supported methods are via the signaling stream using SIP INFO messages or via a DTMF packet that is in compliance with RFC 2833, RTP Payload for DTMF Digits, Telephony Tones and Telephony Signals. Using this object, you can configure the ME to:
Forward packets in the form they arrived.
Pick them out of the RTP stream and send them in a SIP INFO message (if they arrived as DTMF packets).
Extract them from a SIP INFO message and send them as DTMF packets (if the SIP INFO message contained a DTMF body).
Outbound DTMF translation applies to the segment from the ME to the call recipient.
config vsp default-session-config out-dtmf-translation config vsp policies session-policies policy name rule name session-config out-dtmf-translation config vsp dial-plan dial-prefix entryName session-config out-dtmf-translation config vsp dial-plan route name session-config out-dtmf-translation config vsp dial-plan source-route name session-config out-dtmf-translation config vsp session-config-pool entry name out-dtmf-translation
info: Specifies the method to use for forwarding DTMF tones that were received in a SIP INFO message. If set to info, the system forwards the message as it was received (in an INFO message). If set to rfc-2833, the system extracts the DTMF body from the INFO message and sends the content via DTMF packets.
Default: info
Values: info | rfc-2833
Example: set info rfc-2833
drop-info: Specifies whether to drop the SIP INFO message if the info property is set to rfc-2833. If set to true, the system drops the INFO packet and only sends the DTMF packets. If set to false, the system sends both.
Default: false
Values: true | false
Example: set drop-info true
rfc-2833: Specifies the method to use for forwarding DTMF tones that were received in DTMF packets. If set to rfc-2833, the system forwards the message as it was received (in DTMF packets). If set to info, the system extracts the DTMF packets from the RTP stream and sends the content via a SIP INFO message. The system sends one INFO message per event detected.
Default: rfc-2833
Values: info | rfc-2833
drop-rfc-2833: Specifies whether to drop the DTMF packets if the rfc-2833 property is set to info. If set to true, the system drops the RFC 2833 packets and only sends the SIP INFO message. If set to false, the system sends both.
Default: false
Values: true | false
Example: set drop-rfc-2833 true
info-dtmf-body: Specifies the body type to use in a SIP INFO message when converting from RFC 2833 format. When using dtmf, the message body contains just single character (the digit that was pressed). When set to dtmf-relay, the body contains the single character plus duration data.
Default: dtmf-relay
Values: dtmf-relay | dtmf
Example: set info-dtmf-body dtmf
timeout-rfc-2833: Secondary property. Sets the number of milliseconds the system waits before sending a SIP INFO message if it does not detect the end of the event. The timer is started at the start of an event. This property only applies when the forwarding method has been changed from rfc-2833 to info, and is used in the event that when monitoring DTMF, the system does not detect an event end.
Default: 1000
Example: set timeout-rfc-2833 1500
Configures outbound server-side acousting echo cancellation for devices running telephony applications that do not have functional on-board echo cancellation.
config vsp session-config-pool entry name in-echo-cancellation-settings
config vsp default-session-config in-echo-cancellation-settings
admin: Enables or disables echo cancellation on outbound call-legs.
Default: disabled
Values: enabled | disabled
Example: set
platform-delay: Although the maximum echo tail allowed by the echo cancellation algorithm is 200 ms, it is possible that the actual echo tail for certain devices is longer. This parameter allows you to account for any extra delay in addition to the 200 ms. For example, if your total acoustic delay is 500 ms, set echo-tail to 200 ms and platform-delay to 300 ms. This covers the entire 500 ms delay.
Default: 0
Values: Min: 0 / Max: 65535
Example: set platform-delay 100
echo-tail: The amount of time, in milliseconds, that it takes for the person speaking to hear his or her own echo. This is also known as echo delay. The maximum echo tail supported by the ME's algorithm is 200 ms.
Default: 16
Values: Min: 0 / Max: 200
Example: set echo-tail 150
noise-reduction: Allows you to reduce background noise. A value of 0 means there is no noise reduction and 5 is the maximum reduction.
Default: 2
Values: Min: 0 (no noise reduction) / Max: 5
Example: set noise-reduction 4
See the in-hold-translation object for a complete description.
config vsp default-session-config out-hold-translation config vsp policies session-policies policy name rule name session-config out-hold-translation config vsp dial-plan dial-prefix entryName session-config out-hold-translation config vsp dial-plan route name session-config out-hold-translation config vsp dial-plan source-route name session-config out-hold-translation config vsp session-config-pool entry name out-hold-translation
This object determines the ToS value setting for the out-leg of the session. This ToS value determines the quality of service that the call receives. You can either preserve the ToS value in the first received message of the session or overwrite the ToS field of all packets sent out on the out-leg with the value you specify. Enter a number that represents the 8-bit Differentiated Services (DS) field of the IP packet in decimal format, such as 26 for 00011010 or 104 for 01101000. This value can be of use to upstream devices.
config vsp default-session-config h323-tos-settings out-leg-tos config vsp session-config-pool entry h323-tos-settings out-leg-tos
value: Specify the ToS value setting for the out-leg of the session.
Default: preserve; if you select overwrite, the default value is 0
Values: preserve: The ME uses the ToS value in the first received message if the session.
overwrite <value>: The ME marks the ToS field of all packets it sends out on the out-leg with the value you specify.
Example: set value overwrite 22
Changes the media descriptor string (e.g., the CODEC for audio or video) in the SDP. Use this in cases where a client is unable to understand a variation in name of a CODEC/media descriptor. For example, G729 is sometimes transmitted as G729a. Outbound media normalization applies to the segment from the ME to the call recipient.
config vsp default-session-config out-media-normalization config vsp policies session-policies policy name rule name session-config out-media-normalization config vsp dial-plan dial-prefix entryName session-config out-media-normalization config vsp dial-plan route name session-config out-media-normalization config vsp dial-plan source-route name session-config out-media-normalization config vsp session-config-pool entry name out-media-normalization
normalize: Specifies, for a media type, how to normalize a CODEC/media descriptor name. The initial subtype is the type the system matches on and replaces. The alternate subtype is the type that the system then inserts in the SDP to replace the initial subtype. You can select a pre-configured type or enter a custom type.
Default: audio Values: <audio | video | application | image | custom-mime-type mimeType> [initialSubType] [alternateSubType]
Example: set normalize video g729 g729a
Configure out-media-scanner-settings. When in-media-scanner-settings are configured, a media scaner is started on the out-leg of the call and reports events based on the analysis of the received audio from the endpoint.
config vsp default session-config out-media-scanner-settings config vsp session-config-pool entry <name> out-media-scanner-settings config vsp policies session-policies policy <name> rule <name> session-config out-media-scanner-settings
admin: Enables or disables the in-media scanner settings.
Default: disabled
Values: enabled | disabled
Example: set admin enabled
pre-scan-time: The number of milliseconds to delay before invoking the in-media scanner. This property is not applicable for the on-demand media scanner.
Default: 20
Values: Min: 0 / Max: 4294967295
low-threshold: Enter the talk or stable tone signal power threshold in dbs.
Default: -36
Values: Min: -36 / Max: 3
Example: set low-threshold -25
high-threshold: Enter the quiet signal power threshold in dbs. Crossing this threshold indicates talking or stable tone.
Default: -36
Values: Min: -36 / Max: 3
Example: set high-threshold -25
low-long-duration: The number of milliseconds of detected quiet before reporting a long-pause, otherwise a short pause is reported.
Default: 2000
Values: Min: 0 / Max: 4294967295
Example: set low-long-duration 1500
high-long-duration: The number of milliseconds of detected talk or tone before reporting a long talk or stable tone, otherwise a short talk is reported.
Default: 900
Values: Min: 0 / Max: 4294967295
Example: set high-long-duration 1500
averaging-window: Secondary property The amount of time in dbs used when calculating signal strength.
Default: 100
Values: Min: 10 / Max: 1000
Example: set averaging-window 500
nominal-rounding-factor: Secondary property. The signal strength is rounded to the nearest multiple of the value for this property before comparing against other signal strengths.
Default: 2
Values: Min: 1 / Max: 25
Example: set nominal-rounding-factor 4
event-report-frequency: The number of milliseconds the media scanner should wait between generation of media scanner events. Setting this property to 0 causes the media scanner events to be reported immediately as they occur.
Default: 1000
Values: Min: 0 / Max: 60000
Example: set event-report-frequency 39
event-report-count-threshold: The maximum number of media scanner events that can be pended for waiting for the event-report-frequency timer to expire before being reported. If the number of queued media scanner events reaches this count, all of the events will be immediately reported.
Default: 25
Values: Min: 1 / Max: 6000
Example: set event-report-count-threshold 73
event-report-flags: Set the media scanner events to report. The options are:
Default: report all events
Values: short-pause, long-pause, short-talk, long-talk, stable-tone
Example: set event-report-flags short-pause
Configures out-leg MSRP interworking.
config default-session-config out-msrp-session-leg
config session-config-pool entry <name> out-msrp-session-leg
admin: Enable or disable MSRP interworking on this call leg.
Default: disabled
Values: enabled | disabled
Example: set admin enabled
msrp-leg-transport: Specify the MSRP transport method for RCS or WebRTC.
Default: TCP
Values: TCP | TLS | WS | WSS
Example: set msrp-leg-transport TLS
connection-reuse: Not supported in this release.
default-media-interface: Specify the local media interface to use for an MSRP connection if svc-routing fails to locate the appropriate interface.
Default: There is no default setting
Example: set default-media-interface int1
use-mdesc-cline-first: (Advanced) Specify whether the MSRP session manager attempts to use the SDP c-line (before using the path attribute) to learn the remote MSRP endpoint's media IP address.
Default: false
Values: true | false
Example: set use-mdesc-cline-first false
socket-read-size: (Advanced) Specify the MSRP socket read size to use when assembling incoming MSRP messages.
Default: 4096
Min: 0 / Max: 4294967296
Example: set socket-read-size 5000
partial-forward-size: (Advanced) Specify the threshold for forwarding buffered MSRP message content bytes.
Default: 1024
Min: 0 / Max: 4294967296
Example: set partial-forward-size 2050
connsrc-match-path: (Advanced) Specify whether the ME allows incoming MSRP connections even when a remote address does not match the SDP path attribute.
Default: false
Values: true | false
Example: set connsrc-match-path false
allow-missing-fingerprint: (Advanced) Specify whether the ME allows an MSRP secure connection even when the SDP fingerprint attribute is missing.
Default: false
Values: true | false
Example: set allow-missing-fingerprint false
Customizes call admission control, on a per-AOR basis, for outbound calls only. This is in contrast to the admission control settings found in the location-call-admission-control object, which sets aggregate inbound and outbound limits. Use the show location-cac status provider to view call admission control settings and counters for an AOR.
config vsp default-session-config location-call-admission-control outbound-controls config vsp policies session-policies policy name rule name session-config location-call-admission-control outbound-controls config vsp dial-plan dial-prefix entryName session-config location-call-admission-control outbound-controls config vsp dial-plan route name session-config location-call-admission-control outbound-controls config vsp dial-plan source-route name session-config location-call-admission-control outbound-controls config vsp session-config-pool entry name location-call-admission-control outbound-controls
max-number-of-concurrent-calls: Specifies the maximum number of active outgoing calls allowed for this AOR at one time. When this value is reached, the connection does not accept calls until the value drops below the threshold.
A value of 0 causes the system to decline all calls and registrations.
Default: 1000
Values: Min: 0 / Max: 1000000
Example: set max-number-of-concurrent calls 1500
max-calls-in-setup: Sets the maximum number of simultaneous outbound call legs in setup stage that are allowed for this AOR. A call leg in setup is much more compute-intensive than established call legs, so this value is more restrictive than the concurrent call leg value. A value of 0 causes the system to decline all calls and registrations.
Default: 30
Values: Min: 0 / Max: 10000
max-bandwidth: Secondary property. Specifies the amount of bandwidth the system allocates to outbound calls to the AOR. When the system reaches the maximum bandwidth limit for a server, it rejects calls until bandwidth use drops below the maximum.
Default: unlimited Values: unlimited | kbps
Example: set max-bandwidth 512
call-rate-limiting: Secondary property. Limits the number of calls sent from the AOR within a certain interval. Once this interval is reached, the system rejects any calls from this AOR, returning a response code and message until the rate decreases. This feature sets the acceptable set-up rate for incoming calls.
If enabled, set the number of calls allowed and the measurement interval (in seconds). You can also enter a result code from 400 to 699 and a text string to accompany call rejection if no available server is found.
Default: disabled; if set to enabled, the default calls-per-interval is 60, the default interval is 1, and the default result is 486, Busy Here Values: enabled [calls-per-interval][interval][result-code][result-string] | disabled
Example: set call-rate-limiting enabled 50 1 480 ”Temporarily unavailable”
Configures events for outgoing SIP messages. This is a secondary object.
config vsp session-config-pool entry name event-settings outbound-sip-messages
config vsp default-session-config event-settings outbound-sip-messages
admin: Enables or disables events for outgoing SIP messages.
Default: disabled
Values: enabled | disabled
Example: set admin enabled
apply-to-methods-for-events: Select the SIP methods you want the ME to create events for.
Default: There is no default setting
Example: set apply-to-methods-for-events INVITE
apply-to-responses: Specifies whether to send events for SIP requests or both requests and responses. When no, changes are applied to requests only. When yes, changes are applied to requests and responses. If yes, you must set the response code to which it applies. A value of 0 implies all responses.
Default: no Values: no | yes <response code>
Example: set apply-to-responses yes 0
apply-to-dialog: Specifies whether to send events for SIP messages on a specific dialog or not.
Default: both
Values: inbound: Apply to the inbound dialog only
outbound: Apply to the outbound dialog only
both: Apply to both the inbound and outbound dialogs.
Example: set apply-to-dialog inbound
cseq: Secondary property. Sets a mechanism to further filter for which SIP messages have events sent. When set to 0 (the default), the ME sends events for all SIP messages. If set to any other value, the ME only sends events for SIP messages having a CSEQ field that matches that value.
Default: 0
Values: Min: 0 / Max: 4294967296
Example: set cseq 100
Allows the ME to override the dial plan route once a configurable threshold has been reached. Under this object you can configure the session limit and the peer to use once the limit has been reached.
admin: Enables or disables the overflow route feature. When enabled, the ME overrides the dial plan route once a configurable threshold has been reached.
Default: enabled
Values: enabled | disabled
Example: set admin enabled
limit: Enter the session limit threshold to be reached before the overflow route overrides the dial plan route.
Default: 0
Values: Min: 0 / Max: 20000
Example: set limit 150
peer: Specify the peer to which you want traffic forwarded once the session limit threshold has been reached. Enter the peer type and address.
Default: none
Values: none
server
carrier
exchange
switch
trunk
hunt-group
calling-group
virtual-dial-plan
Example: set peer server ”vsp\enterprise\servers\sip-gateway RedirectClusters
Specifies what derives the content of the fields of the outgoing P-Asserted-Identity URI when the ME transmits a message. For example, if the user property is set to to-uri, the system replaces the user field of the P-Asserted-Identity URI with data from the user field of the incoming TO URI. If set to omit, the user field is left blank. Or, you can enter any string that you want placed in the user field. See Altering URIs for information on replacement options.
config vsp default-session-config p-asserted-identity- uri-specification config vsp policies session-policies policy name rule name session-config p-asserted-identity-uri-specification config vsp dial-plan dial-prefix entryName session-config p-asserted-identity-uri-specification config vsp dial-plan route name session-config p-asserted-identity-uri-specification config vsp dial-plan source-route name session-config p-asserted-identity-uri-specification config vsp session-config-pool entry name p-asserted-identity- uri-specification
user: Specifies how to derive the value of the user field of the P-Asserted-Identity URI.
Default: same-uri Values: same-uri | request-uri | to-uri | from-uri | omit | next-hop | string
Example: set user from-uri
host: Specifies how to derive the value of the host field of the P-Asserted-Identity URI.
Default: same-uri Values: same-uri | request-uri | to-uri | from-uri | omit | next-hop | next-hop-domain | local-ip | string
Example: set host request-uri
port: Specifies how to derive the value of the port field of the P-Asserted-Identity URI.
Default: same-uri Values: same-uri | request-uri | to-uri | from-uri | omit | string
Example: set port omit
display: Specifies how to derive the value of the display field of the P-Asserted-Identity URI.
Default: same-uri Values: same-uri | request-uri | to-uri | from-uri | omit | next-hop | string
Example: set display to-uri
transport: Specifies the value of the transport field of the P-Asserted-Identity URI. In addition to using the value from other fields of the incoming URI, you can set the transport method to UDP, TCP, TLS, or the method used by the next-hop server.
You cannot enter a string for this property.
Default: same-uri
Values: same-uri | request-uri | to-uri | from-uri | omit | UDP | TCP | TLS | next-hop
Example: set transport next-hope
user-param: Specifies whether the User parameter in the P-Asserted-Identity URI of the SIP header is maintained or removed when the system forwards a message. If set to keep, the message is forwarded with the parameter as it was received. If set to omit, the entire user=param is removed from the P-Asserted-Identity URI.
Default: omit
Values: omit | keep
Example: set user-param keep
uri-parameter <name><value>: Appends the specified user parameter and value to the P-Asserted-Identity URI for matching calls. The resulting parameter is added to the URI. For example, the example below would result in a P-Asserted-Identity URI that looked similar to:
<sip:jane@fun.com;BTG=trunk1>
You can append multiple users.
Default: There is no default setting
Example: set uri-parameter BTG trunk1
create: Sets whether to create a P-Asserted-Identity header if one does not already exist. If true, the system uses the property settings in this object to determine the value of the fields in the header. If set to false, the system does not create the header. If a header does already exist, this field has no effect. (The fields of the existing header are still manipulated by the values of the object's properties.)
Default: false
Values: true | false
Example: set create true
use-original-from-header: Specifies where the content of the P-Asserted-Identity header is derived from. (This property is only applicable if the create property is set to true.) When use-original-from-header is true, the system uses the From header as it existed, before any normalization occurred, as the p-asserted-identity header. If set to false, the system uses the From header as it exists when SIP call processing occurs (e.g., post normalization).
Default: false
Values: true | false
Example: set use-original-from-header true
Sets the next-hop destination server where the ME forwards a SIP messages. If this object is not set, calculations merged from the dial plan, location cache, and other policy settings control how the ME forwards a message. These settings override the dynamic calculations, and effect the routing calculation that determines the next-hop server.
config vsp default-session-config peer config vsp policies session-policies policy name rule name session-config peer config vsp dial-plan dial-prefix entryName session-config peer config vsp dial-plan route name session-config peer config vsp dial-plan source-route name session-config peer config vsp session-config-pool entry name peer
peer: Specifies to which server the system should forward the call. Enter the name of a previously configured server or group of the type specified.
Default: automatic
Values: trunk: Enter a reference to a carrier\exchange\switch trunk-group and optionally, a rate plan.
switch: Enter a reference to a carrier exchange dial-plan and optionally, a rate plan.
exchange: Enter a reference to a carrier dial-plan and optionally, a rate plan.
server: Enter a reference to an enterprise dial-plan.
calling-group: Enter a reference to a calling-group dial-plan.
server-member: Enter a reference to a server-pool server-pool-admission-control.
uac: Allows you to specify any device, using the contact string that may be found in the location binding. To use the contact string in the location binding, use the show location-cache -v command. If the contact address is marked as true, it is stored as secure, using a SIPS: indicator.
automatic: Sets the system to automatically route the session. In this case, the system bases routing decisions on settings in other objects (e.g., location cache, dial plan, and other policy settings).
named: Identifies the next-hop server by matching on enterprise server carrier and endpoint tags. If multiple matches occur, the ME hunts through all matches. To identify the server, enter the carrier tag used for an enterprise dial-plan and the endpoint tag used for the server-pool > server-pool-admission-control. If the tags configured here do not match configured tags, no server is set as the next-hop and the ME responds with a 404 (notfound) message.
Example: set peer uac sip:cov2075556789@elmaple.com true
hunt-timeout: Specifies the number of seconds to wait before determining a server is unavailable and trying the next configured server.
Default: 30
Example: set hunt-timeout 25
Specifies a WAV file that will be periodically inserted into a call. Use this to enter a recording tone or hold message. Use the file-play-verify action to ensure that the recording is of a format supported by the ME device.
config vsp default-session-config media periodic-announcement config vsp policies session-policies policy name rule name session-config media periodic-announcement config vsp dial-plan dial-prefix entryName session-config media periodic-announcement config vsp dial-plan route name session-config media periodic-announcement config vsp dial-plan source-route name session-config media periodic-announcement config vsp session-config-pool entry name media periodic-announcement
file: Specifies the location of the system of a WAV file containing the announcement.
Default: There is no default setting
Example: set file /cxc/recordings/announce1
period: Specifies the number of seconds the system waits between insertions of the announcement into the call. Note that the system starts its timer at the beginning of the announcement. The file starts playing every period number of seconds, regardless of the length of the recording. This means that you will hear the announcement continuously if the file, or specified duration (below), are longer than the specified period.
Default: 30
Values: Min: 10 / Max: 3600
Example: set period 45
duration: Specifies, in milliseconds, how much of the specified recording to play. If you specify 0, the system plays the recording in its entirety. Use this with a value set to insert a tone file, and set the number of milliseconds that would not be too intrusive.
Default: 0
Example: set duration 500
Enables or disables playback of the last recorded SIP call from a specific To/ From pair. Playback can be initiated using the dial-prefix object or any other type of policy that would trigger this object through the session configuration.
Note:
To use this feature, you must enable the anchor and record properties in the media object under the default session configuration.For example, you could configure the dial-prefix to recognize *73. When a call came in with that prefix in the To or From URI of the SIP header, it would trigger the session configuration associated with that dial prefix plan. With this object enabled, the *73 would initiate playback of the last call. Therefore,
To:*73bob@phone.com
From:joe@phone.com
results in the ME playing back, to joe@phone.com, the last call exchange that was from joe@phone.com and to bob@phone.com. The ME searches the database for this call and initiates a call back to SIP phone joe@phone.com, playing back the recording of the previous call instead of connecting a new one.
You can also configure the ME to play back the last call recorded call from a phone, regardless of the destination. For joe@phone.com to hear his last recorded call, he would initiate a call to himself using the configured dial-prefix (*73joe@phone.com).
config vsp default-session-config playback-call-settings config vsp policies session-policies policy name rule name session-config playback-call-settings config vsp dial-plan dial-prefix entryName session-config playback-call-settings config vsp dial-plan route name session-config playback-call-settings config vsp dial-plan source-route name session-config playback-call-settings config vsp session-config-pool entry name playback-call-settings
playback-last-call: Configures the system to playback the last call between the To/From call pair instead of initiating a new call. Note that the anchor and record properties must be enabled in the media object of the default session configuration for recording to take place.
Default: disabled
Values: enabled | disabled
Example: set playback-last-call enabled
Configures the call access mechanism in the ME. When configured and enabled, the ME terminates matching calls at the box. The system then collects the DTMF digits it receives until either the timeout expires, the verify signal is received, or maximum number of digits has been reached. After digit collection is terminated, the ME compares the digits to its list of authorized numbers for that caller. If the collected digit string matches an entry in that list, the ME plays a success message, if configured, and completes the call. If the collected digit string does not match an entry in that list, the ME plays the failure message, if configured, and terminates the call. In addition, while digits are being collected, if the ME encounters the configured ”cancel” digit, the system terminates the call. If the ”restart” digit is enabled and encountered, all previously entered digits are removed, and collection continues.
config vsp default-session-config pre-call-authorization config vsp policies session-policies policy name rule name session-config pre-call-authorization config vsp dial-plan dial-prefix entryName session-config pre-call-authorization config vsp dial-plan route name session-config pre-call-authorization config vsp dial-plan source-route name session-config pre-call-authorization config vsp session-config-pool entry name pre-call-authorization
admin: Sets the administrative status of the configuration entry for terminating incoming calls at the box.
Default: disabled
Values: enabled | disabled
Example: set admin enabled
greeting-message: Sets the message that is played to incoming calls if they have been terminated. Enter a path name to the WAV file containing the message.
Default: There is no default setting
Example: set greeting-message /cxc_common/preauth.wav
max-number-of-digits: Specifies the maximum number of digits the ME collects before it begins evaluating the string against the authorized list (set with the authList property).
Default: 7
Values: Min: 1 / Max: 16
Example: set max-number-of-digits 10
verify: Configures a signal (telephone key pad key) that the ME waits for before it begins evaluating the string against the authorized list (set with the authList property). Use this to allow, for example, variable length PINs. Configure a pound (#) symbol to indicate end of collection and end PIN entries with a pound.
Default: disabled Values: enabled telephoneKey | disabled
Example: set verify enabled Pound
cancel: Configures forced termination of a call. When enabled, sets a signal (telephone key pad key) that, if encountered in DTMF collection, forces the ME to terminate the call.
Default: disabled Values: enabled telephoneKey | disabled
Example: set cancel enabled Star
restart: Configures a collection restart. When enabled, sets a signal (telephone key pad key) that, if encountered in DTMF collection, forces the ME to wipe whatever digits have been collected and restart collection.
Default: disabled Values: enabled telephoneKey | disabled
Example: set restart enabled Flash
inter-digit-timeout: Configures a length of time the ME waits after button pushing has stopped and before it begins evaluating the string against the authorized list (set with the authList property). The inter-digit timer is reset every time a digit is received. When it times out, the preceding digits will be verified. This is used, for example, when entering a PIN. The number is not verified until the caller stops entering; after the inter-digit time out expires the ME checks the string.
Default: 2500
Values: Min: 500 / Max: 60000
Example: set inter-digit-timeout 5000
success-message: Sets the message that is played to the caller indicating that the call attempt has been successful. This is played if the collected digits match an entry in the authorized list (set with the authList property). Enter a path name to the WAV file containing the message.
Default: There is no default setting
Example: set success-message /cxc_common/callSuccess.wav failure-message: Sets the message that is played to the caller indicating that the call attempt was not successful. This is played if the collected digits do not match an entry in the authorized list (set with the authList property). Enter a path name to the WAV file containing the message.
Default: There is no default setting
Example: set failure-message /cxc_common/callFail.wav
authList: Adds strings of digits to a list of authorized calling numbers. The ME uses this list to match against the collected digits and determine whether a call is allowed. Re-execute the command to enter multiple strings.
Default: There is no default setting
Example: set authList 011
Configures presence services for this SIP call session. SIP call messages containing SUBSCRIBE and NOTIFY requests and responses are associated with a presence session.
The ME stores presence information in its local databases. Presence information pertains to your SIP presence (online, away, etc.). This object enables presence services and controls whether the ME translates that presence information from one server type to another (because every server type has its own way of storing/transmitting this data).
config vsp default-session-config presence config vsp policies session-policies policy name rule name session-config presence config vsp dial-plan dial-prefix entryName session-config presence config vsp dial-plan route name session-config presence config vsp dial-plan source-route name session-config presence config vsp session-config-pool entry name presence
presence-services: Enables or disables system presence services on this SIP call session. If set to enabled, the session information is sent to the system presence database. If disabled, the system allows packets through but does not write any information to the database.
Default: disabled
Values: enabled | disabled
Example: set presence-services enabled
presence-translation: Sets the server types, and therefore the translation direction, that the system performs. The system modifies an incoming request so that the far-end client receives presence data in a recognizable format.
Default: disabled
Values: disabled: The request is passed through unmodified
sametime-to-lcs: Provides Sametime-to-LCS translation
lcs-to-sametime: Provides LCS-to-Sametime translation
lcs-to-lcs: Ensures LCS-to-LCS presence data compatibility.
st-to-st: Ensures Sametime-to-Sametime presence data compatibility. This would be for use, for example, when using different versions of Sametime.
mcs-to-mcs: Ensures Nortel MCS-to-MCS presence data compatibility, for example, when different versions are running.
voice-to-voice: Indicates that the entity is a phone/voice (to support phones that want to SUBSCRIBE to a SIP server and announce presence). Identifying the entity as voice prevents the system from altering the SUBSCRIBE message in the presence database.
voice-to-lcs: Enables forwarding of phone registrations to LCS (mostly useful in CSTA cases) to control OC client presence information.
Example: set presence-translation sametime-to-lcs presence-mapping <value><mapTo>: Configures, for a given presence status, the value to map that status to. Use this property if both clients do not recognize the same presence states. In an LCS-to-Sametime configuration, for example, if status is set to a value of ”Out to Lunch” with the LCS client, it must be mapped to an option that Sametime recognizes.
Default: There is no default setting
Values: offline | away | out-to-lunch | on-the-phone | be-right-back | busy | do-not-disturb | online
Example: set presence-mapping out-to-lunch away
voice-lcs-transport: Sets the default protocol used to transmit voice-to-LCS SIP packets to the destination LCS server.
Default: any
Values: any | UDP | TCP | TLS
Example: set voice-lcs-transport UDP
federation-contact: Specifies the string that the system uses to create the SIP header URIs it sends to remote servers. The string must match the content of the Common Name field that is present in the certificate that the system provides to the LCS to prove its authenticity.
Default: There is no default setting
Example: set federation-contact companyABC.com
message-body-replace: Alters the body of any SIP message (for example, IM content or the SDP for an INVITE) for a matching session. Use this property with caution; you would only change the SIP message body under specific required circumstances.
In the example below, the system replaces sip:1002@company.com in the SIP message body with sip:9788231002@company.com.
For more information regarding configuring regular expressions and replacement strings, see Using Regular Expressions.
Default: There is no default setting
Example: set message-body-replace ”(.*)sip:(\d{4})@company.com(.*)””\1sip:978823\2@company.com\3”
st-keep-alives: Specifies whether the system responds to Sametime keep alives. When enabled, the system responds to keep alives on behalf of the federated domain, letting the Sametime server know that the remote peer is active. Use this in federated configurations using a version of Sametime that requires SIP keep alives. Otherwise, this property should be disabled.
Default: disabled
Values: enabled | disabled
Example: set st-keep-alives enabled
location-presence-service: Specifies whether the ME publishes device presence information to the jtapi master service. For Oracle Technical Support use only.
Default: disabled
Values: enabled | disabled
Example: set location-presence-service enabled
The provisional-response object allows you to add any additional provisional responses you want sent along after the ”100 Trying” response at the start of a normal INVITE dialog.
Allows the configuration of q931-cause and/or H.225 reason code for calls cleared by an external SIP UA. When an IW call is cleared on the SIP side, the SIP response code is used to consult an internal table for Q.931/H.225 information needed when generating the ReleaseComplete, Admission Reject, or Location Reject message. By adding a q931-cause-sip-response-map entry, you can override the internal table defaults.
config vsp session-config-pool entry q931-cause-sip-response-map config vsp default-session-config q931-cause-sip-response-map
q931-cause: Select a q931-cause to use when clearing the H.323 side of the call. If this map entry will not generate a q931-cause, or you want to use the default, select Any.
Default: any
Example: set q931-cause userby
h2250-reason: Select a h225-reason to use when clearing the H.323 side of the call.
Default: none
Values: none: Use the default h225-reason
lrj: Select lrj if you are generating LRJ messages and enter a relevant reason
arj: Select arj if you are generating ARJ messages and enter a relevant reason
any: Specifying only the q931-cause in this entry
Example: set h2250-reason lrj
sip-response: Select the sip-response match criteria for this entry. If this entry will not generate a q931-cause or you want to use the default, select Any.
Default: 0
Values: Min: 300 / Max: 699
Example: set sip-response 500
The q931-settings configuration object is used to configure Q.931 settings on the ME.
config vsp session-config-pool entry q931-settings config vsp default-session-config q931-settings
numbering-plan: The Q.931 numbering plan set in Calling and Called Party number information elements.
Default: ISDN
Values: unknown | ISDN | data | telex | national-standard | private | reserved
Example: set numbering-plan data
numbering-type: The Q.931 numbering type set in Calling and Called Party number information elements.
Default: international
Values: unknown | international | national | network-specific | subscriber | abbreviated | reserved
Example: set numbering-type national
presentation-indicator: Enter the static presentation value to use.
Default: allowed
Values: allowed | restricted | numberNotAvailable | reserved
Example: set presentation-indicator restricted
screening-indicator: Enter the static screening value to use.
Default: notScreened
Values: notScreened | verifiedPassed | verifiedFailed | networkProvided
Example: set screening-indicator verifiedPassed
privacy-dynamic: When true, the screening and presentation are dynamic.
Default: true
Values: true | false
Example: set privacy-dynamic false
use-display-ie: When true, the ME attempts to use Display IE from the SETUP message when building SIP From: header display-name.
Default: true
Values: true | false
Example: set use-display-ie false
add-outgoing-displaytext-ie: When true, the ME attempts to use SIP From: header display-name when building Display IE in the outgoing SETUP message.
Default: false
Values: true | false
Example: set add-outgoing-displaytext-ie true q931-bearer-capability-ie [coding-standard][info-transfer-capability][transfer-mode][transfer-rate][user-info-layer1-protocol]: Sets the Q.931 Bearer Capability values used in outgoing H.323 messages.
Example: set q931-bearer-capability-ie nationalStd speech packetMode 128Kbps h261
Specifies whether to record calls, and if so, how to handle unsupported CODECs. Note that if you enable the record property, enabling recording of the SIP session, you must also have enabled the anchor property of the media object.
If the record property is enabled, you can use CLI actions or the ME Management System to replay the SIP call session. To use the CLI, use the mix-session action to mix the files manually. You can also:
play the file using the file-play action.
play the session using the playback session action.
When the ME receives the SDP message (typically in either an INVITE or 200 OK), it contains information from the originator on supported CODECs. These CODECs may or may not be mixable by the ME device. When the strip-unplayable property is enabled, the ME removes any CODECs that it cannot mix from the list, and forwards the INVITE to the destination. When disabled, the ME forwards the INVITE with the CODEC list unchanged. The ME then records the entire contents of the call, but when mixed, any data sent with an unsupported CODEC results in silence.
Note:
If The ME records a call containing an unplayable CODEC, any archiving operation that involves that record (because the include-mixed-media property of the archiving operation is set to true) will fail.config vsp default-session-config media recording-policy config vsp policies session-policies policy name rule name session-config media recording-policy config vsp dial-plan dial-prefix entryName session-config media recording-policy config vsp dial-plan route name session-config media recording-policy config vsp dial-plan source-route name session-config media recording-policy config vsp session-config-pool entry name media recording-policy
record: Enables or disables recording of the SIP session to the default directory on the system disk system. See Playing Recorded Calls for more information.p
Default: disabled
Values: enabled | disabled
Example: set record enabled
strip-unplayable: Specifies whether or not to strip CODECs that the system does not support. (This is only applicable if recording is enabled.) See Handling Unplayable CODECS for more information, specifically about the risk to archiving operations if set to disabled.
Use the file-play-verify action to ensure that a recording is of a format supported by the ME device.
Default: enabled
Values: enabled | disabled
Example: set strip-unplayable disabled
Enables or disables call parking compatibility settings for the Sylantro SIP for Business initiative.
config vsp default-session-config refer-settings config vsp policies session-policies policy name rule name session-config refer-settings config vsp dial-plan dial-prefix entryName session-config refer-settings config vsp dial-plan route name session-config refer-settings config vsp dial-plan source-route name session-config refer-settings config vsp session-config-pool entry name refer-settings
modify-call-parking: Enables or disables call parking compatibility features. When enabled, the system modifies the Refer-To header. Leave this property at the default setting of disabled if you are not using the SIP for Business platform.
Default: disabled
Values: enabled | disabled
Example: set modify-call-parking enabled
fix-refer-to-call-id: Specifies whether to modify the call ID value in the Refer-to header. Enable this property if there are problems, caused by an incorrectly implemented NAT device, with the Replaces section of the Refer-to header. When enabled, if the system detects a @ipAddress string at the end of the Refer-to header call-id field, the system replaces the address with the message's remote IP address. This allows the system to later use that value to find the call leg it refers to and perform a correct translation.
Default: disabled
Values: enabled | disabled
Example: set fix-refer-to-call-id enabled
Provides granular modification capabilities to create or modify a header. Use this when the methods of the altered-header object do not achieve the necessary changes. You can create multiple reg-ex-header entries to accomplish complicated manipulations. The system executes each one, the order of execution based on their order within the configuration.
config vsp default-session-config header-settings reg-ex-header number config vsp policies session-policies policy name rule name session-config header-settings reg-ex-header number config vsp dial-plan dial-prefix entryName session-config header-settings reg-ex-header number config vsp dial-plan route name session-config header-settings reg-ex-header number config vsp dial-plan source-route name session-config header-settings reg-ex-header number config vsp session-config-pool entry name header-settings reg-ex-header number
admin: Enables or disables this configuration entry.
Default: enabled
Values: enabled | disabled
Example: set admin disabled
destination: Specifies the header to be created or modified by the properties set in this object (the header to write the changes to). If the header doesn't exist in the message, the system creates it. When you configure this, you must also set either the create or append properties. Otherwise, there is no configuration from which to derive the new URI. You do not specify the fields within the destination to change as that level of change is accomplished with the regular expression statements. To modify an existing header, use the same header for the both the destination and the create properties.
Default: There is no default setting
Example: set destination Diversion
create <expression><replacement>: Identifies the header containing the data to be modified and then written to the destination header. (Use the append property to add data to the existing header.) First, select the header that serves as the source of the data. Then, specify a regular expression to run against the value of the source header. Also supply the replacement expression to apply if there's a match. Changes are only applied to the original expression match.
For more information regarding configuring regular expressions and replacement strings, see Using Regular Expressions.
Default: There is no default setting
Example: set create History-Info "([^,<]*?)<sip:([^>?]*?)\?Reason=SIP%3Bcause%3D3[0-9]{2}(.*?)>(.*?)index=(\d.\d.\d|\d.\d|\d)" "<sip:\2>;reason=deflection"
append<expression><replacement>: Identifies the header containing the data to be added to the destination header. First, select the header that serves as the source of the data. Then, specify a regular expression to run against the value of the source header. Also supply the replacement expression to apply if there's a match. The system appends this string to the existing destination header. To add spaces or commas, be sure to include them (using quotation marks) in the replacement statement.
For more information regarding configuring regular expressions and replacement strings, see Using Regular Expressions.
Default: There is no default setting
Example: set append History-Info ([^,<]*?)<sip:cov.com ", <sip:cov2.com"
apply-to-methods: Specifies the message type to which the system applies header value changes. The system then changes the specified URI according to the settings of the header and destination properties of this object.
When you modify this value, the system overwrites the current setting with only the message types you specify. For example, if set to the default and you enter INVITE, the system only authenticates INVITE messages. Enter multiple types separated by a plus sign (+) with no spaces.
Default: INVITE
Example: set apply-to-methods INVITE+REFER
apply-to-responses: Specifies whether to apply header value changes to SIP requests or requests and responses. Set to no to apply changes only to requests. Set to yes to apply to responses as well. If yes, you must set the response code to which it applies. Create additional altered-body profiles to change multiple response types.
Default: no Values: no | yes responseCode
Example: set apply-to-responses yes 200
session-persistent: Specifies to which messages in a session the ME should apply changes made with this object. When enabled, the ME applies any TO, FROM, or REQUEST URI changes to the first and all subsequent messages in a session. When disabled, the default, the system applies the changes only to the first message in the session.
Default: disabled
Values: enabled | disabled
Example: set session-persistent enabled
cseq: Secondary property. Sets a mechanism to further filter which SIP messages have the header expression modifications applied. If cseq is set to zero (the default), the ME applies the changes to all SIP messages. If set to any other value, the system only applies the changes to SIP messages having a CSEQ field that matches that value.
Default: 0
Example: set cseq 100
apply-to-dialog: Allows you to configure where to apply these options for a session.
Default: both
Values: inbound: Apply to inbound dialog only.
outbound: Apply to outbound dialog only.
both: Apply to both inbound and outbound dialogs.
Example: set apply-to-dialog inbound
create-on-failed-match: Secondary property. When true, construct a create header even when the expression is not a complete match.
Default: true
Values: true | false
Example: set create-on-failed-match false
append-on-failed-match: Secondary property. When true, execute the append action event when the create expression fails to match.
Default: true
Values: true | false
Example: set append-on-failed-match false
Configures properties that are used in processing REGISTER requests. Note that for the ME to accept REGISTER requests, you must enable the admin property of the registration-service object.
config vsp default-session-config registration config vsp policies session-policies policy name rule name session-config registration config vsp dial-plan dial-prefix entryName session-config registration config vsp dial-plan route name session-config registration config vsp dial-plan source-route name session-config registration config vsp session-config-pool entry name registration
admin: Specifies whether to accept or deny REGISTER requests. If set to enabled, the system accepts REGISTERs; the system denies REGISTERs if set to disabled.
Default: disabled
Values: enabled | disabled
Example: set admin enabled
cache-lcs: Specifies whether the system should add entries for LCS clients to the registration database. Use this object in single-domain configurations, where the system sits between the client and the LCS server. When enabled, client requests that pass through the system are added to the database. The system adds an entry for the LCS client record into its registration database. Therefore, you can enable features that use the database, such as call forking.
When disabled, only client requests from external domains are added to the database. The system passes all call requests directly to the LCS server.
Default: disabled
Values: enabled | disabled
Example: set cache-lcs enabled
add-path-header: Specifies whether the system adds a Path: header to the packet. The Path header, with syntax similar to the Record-Route header, is used in conjunction with REGISTER requests and with 2xx messages.
Default: disabled
Values: enabled | disabled
Example: set add-path-header enabled
local-options-reply: Specifies how the system responds to OPTIONS requests during a registration session. When enabled, the system flags the matching AOR entry in the location cache with this setting. In future registration sessions for that AOR, the system will respond locally to any OPTIONS requests, rather than forwarding them to the phone. When disabled, OPTIONS requests are forwarded normally.
Default: disabled
Values: enabled | disabled
Example: set local-options-reply enabled
handle-3xx-locally: Specifies whether the system forwards responses to 3xx messages back to the UAC or resends them. If disabled, when the system receives a 3xx response for a REGISTER (e.g., 301 Moved Permanently or 302 Moved Temporarily), it forwards the response back to the client. When this option is enabled, the system does not forward the response back. Instead, it hunts through the contact routes in the response message until it finds one that responds with a 200 OK. It then applies that route to the REGISTER message.
Default: disabled
Values: enabled | disabled
Example: set handle-3xx-locally enabled
delegate-aged-bindings: Specifies whether the system can redelegate a REGISTER request before both sides experience a binding expiration. By default, when the binding expires on the client-side, the system does not redelegate the REGISTER until the peer expiration. This can cause problems if, for example, you are using DNS to load balance REGISTERs between systems. In that case, the REGISTER may not be redelegated to the third-party server until the peer timer expires, making the server unaware of the relocation. When enabled, the system redelegates the REGISTER request to the peer even if only the client-side timer has expired.
Default: disabled
Values: enabled | disabled
Example: set delegate-aged-bindings enabled
unregister-aged-bindings: Specifies whether the ME allows delegation of an unregister-all (*) request. When enabled, the ME delegates the Contact: * UNREGISTER request unchanged and removes all bindings associated with the AOR that have this feature enabled.
Default: disabled
Values: enabled | disabled
Example: set unregister-aged-bindings enabled
delegate-unregister-all: Specifies whether the system sends an UNREGISTER request when a binding ages out. When enabled, if a binding ages out the system sends an UNREGISTER to the delegate. The system also resets the peer expiration so that the next REGISTER from this client is delegated. You can remove the binding without sending the UNREGISTER by using the location flush now action.
Default: disabled
Values: enabled | disabled
Example: set delegate-unregister-all enabled
broadsoft-survivability-mode: Secondary property. Enables or disables the Broadsoft survivability feature.
Default: disabled
Values: enabled | disabled
Example: set broadsoft-survivability-mode enabled
broadsoft-registration-survivability-mode: Enables or disables the Broadsoft survivability registration feature. When enabled, enter an expression, a replacement, and the client-expiration. The expression is the value in the extension provided in the 200 OK and the replacement is the value for the ME to use in its place to build the local location cache mapping. The client-expiration is the time, in seconds, that the ME waits to recheck the state of the Broadsoft server.
Default: disabled Values: enabled <expression> <replacement> <client-expiration> | disabled
Example: set broadsoft-registration-survivability-mode enabled ”^sip:.'(\d\d\d\d)@””\1” 30
Configures the settings for header manipulation of the User and/or Host portion of the Remote-Party-ID header of the SIP message. Within this object you can modify the existing header or create a new header if one is not present. You can then modify the content by prepending or stripping off characters, for example, to ensure interoperability between carriers. Typically these headers are only part of INVITE messages.
config vsp default-session-config remote-party-id-specification config vsp policies session-policies policy name rule name session-config remote-party-id-specification config vsp dial-plan dial-prefix entryName session-config remote-party-id-specification config vsp dial-plan route name session-config remote-party-id-specification config vsp dial-plan source-route name session-config remote-party-id-specification config vsp session-config-pool entry name remote-party-id-specification
user-source: Specifies the URI from which the system extracts the User portion and applies it to the Remote-Party-ID header: either From, To, Request, or Contact. If left at the default, remote-party-id-uri, there is no change to the USER portion. For example, if you select from-uri, the system extracts the User portion of the From URI and applies it to overwrite or create the User portion of the existing or newly created Remote-Party-ID header.
Default: remote-party-id-uri
Values: remote-party-id-uri | from-uri | to-uri | request-uri | contact-uri
Example: set user-source from-uri
user-action: Sets the action to take once the system has derived the content of the Remote-Party-ID header. You can prepend, strip-off, or replace digits to the User portion. You can also make no modifications. (For example, you may want to create a header using the remote-party-id-specification object, but not want to make any modifications to the default header that is created.)
Default: none Values: none | prepend phone-prefix | strip-off-number | replace-with new-number
Example: set user-action prepend 1
host-source: Specifies where the system extracts the Host portion for the new or existing Remote-Party-ID header. Set the host field to be derived from the remote-party-id, To, From, Request, or Contact URIs of the INVITE, or you can enter any text string.
Default: remote-party-id-uri Values: remote-party-id-uri | from-uri | to-uri | request-uri | contact-uri | string
Example: set host-source to-uri
create: Sets whether to create a Remote-Party-ID header if one does not already exist. If true, the system uses the property settings in this object to determine the value of the fields in the header. If set to false, the system does not create the header. If a header already exists, this field has no effect. (The fields of the existing header are still manipulated by the property values.
Default: false
Values: true | false
Example: set create true
Specifies what derives the content of the fields of the REQUEST URI when the ME transmits a message. For example, if the user property is set to to-uri, the ME replaces the user field of the REQUEST URI with data from the user field of the incoming TO URI. If set to omit, the user field is left blank. Or, you can enter any string that you want placed in the user field. See Altering URIs for information on replacement options.
config vsp default-session-config request-uri-specification config vsp policies session-policies policy name rule name session-config request-uri-specification config vsp dial-plan dial-prefix entryName session-config request-uri-specification config vsp dial-plan route name session-config request-uri-specification config vsp dial-plan source-route name session-config request-uri-specification config vsp session-config-pool entry name request-uri-specification
user: Specifies how to derive the value of the user field of the REQUEST URI.
Default: request-uri Values: request-uri | to-uri | from-uri | omit | next-hop | local | omit-phone-context | string
Example: set user request-uri
host: Specifies how to derive the value of the host field of the REQUEST URI.
Default: request-uri Values: request-uri | to-uri | from-uri | omit | next-hop | next-hop-domain | local-ip | string
Example: set host omit
port: Specifies how to derive the value of the port field of the REQUEST URI.
Default: request-uri Values: request-uri | to-uri | from-uri | omit | string
Example: set port omit
transport: Specifies the value of the transport field of the REQUEST URI. In addition to using the value from other fields of the incoming URI, you can set the transport method to UDP, TCP, or TLS.
You cannot enter a string for this property.
Default: request-uri Values: request-uri | to-uri | from-uri | omit | UDP | TCP | TLS | string
Example: set transport omit
use-param: Specifies whether the User parameter in the REQUEST URI of the SIP header is maintained or removed when the system forwards a message. If set to keep, the message is forwarded with the parameter as it was received. If set to omit, the entire user=param is removed from the TO URI.
Default: omit
Values: omit | keep
Example: set use-param keep
user-truncate-non-digits: Specifies whether to remove non-digits from the User portion of the REQUEST URI in INVITE messages. When enabled, the system removes all non-digits.
Default: disabled
Values: enabled | disabled
Example: set user-truncate-non-digits enabled
uri-parameter <name><value>: Appends the specified user parameter and value to the REQUEST URI for matching calls. For example, the example below would result in a REQUEST URI that looked similar to:
<sip:spot@fun.com;BTG=trunk1>
You can control how and when the new parameter is added using the append options. Select append-always to have the parameter added to all matching calls. Select append-if-does-not-exist to add the name and value only if it does not already exist in the URI, preventing the possibility of duplicate parameters. Select overwrite-existing to replace any existing parameter in the REQUEST URI with the configured name and value, updating instead of appending to the parameter. You can append multiple user parameters.
Default: There is no default setting
Example: set uri-parameter BTG trunk1
apply-to-routing: Specifies whether the system uses the REQUEST URI changes that result from the settings for routing and normalization or for normalization only. When set to false, the default, the system only changes the REQUEST URI for the next-hop SIP server (normalization). When true, the system uses the information in the modified REQUEST URI to forward the message (routing), bypassing the functions of the dial plan and/or location cache.
Set this property to true only when the system is operating in proxy mode and relies on the modified REQUEST URI to correctly forward the message. For example, set this property to true in a CSTA application to cause the system to use the new REQUEST URI to resolve routing.
Default: false
Values: true | false
Example: set apply-to-routing true
use-location-cache-contact-uri: Specifies whether the system should use the location cache information or the request-uri-specification configuration to modify the contact field of the REQUEST message. This property is only applicable when the system uses the location service to forward the request. When true, the system copies the phone contact URI (as found in the location cache) to the request URI. When false, the system uses the settings of this object to determine the content of the REQUEST message.
Default: true
Values: true | false
Example: set use-location-cache-contact-uri false
Maps a new status code and, optionally, phrase to a received code.
config vsp default-session-config response-translation-settings config vsp policies session-policies policy name rule name session-config response-translation-settings config vsp dial-plan dial-prefix entryName session-config response-translation-settings config vsp dial-plan route name session-config response-translation-settings config vsp dial-plan source-route name session-config response-translation-settings config vsp session-config-pool entry name response-translation-settings
entry <statusCode><newStatusCode>[reason][newReason]: Sets a new numeric to a reason code. Specify a code number and the replacement number to use instead. Optionally, you can specify the reason phrase and a replacement phrase. If you specify both the code and phrase, the incoming message must contain both for the replacement to take place.
Default: none
Example: set entry 503 200 ”Service Unavailable” ”Service Delayed”
Provides mechanisms for filtering the routing table. For example, when configured and a policy match occurs, the ME can send traffic to interfaces with the same geolocation setting. Or, the ME can control the egress interface using the routing and classification tag system. (See Tag-Based Route Selection for a complete description of how the ME classifies traffic.) Otherwise, the ME uses the entire routing table when identifying interface choices.
Note:
The preferred method for creating virtual firewalls is by using routing tags and VLANs. For sample configurations that illustrate VLANs, overlapping IP addresses, and virtual firewalls, see the Oracle Communications WebRTC Session Controller Administration Guide.config vsp default-session-config routing-settings config vsp policies session-policies policy name rule name session-config routing-settings config vsp dial-plan dial-prefix entryName session-config routing-settings config vsp dial-plan route name session-config routing-settings config vsp dial-plan source-route name session-config routing-settings config vsp session-config-pool entry name routing-settings
admin: Enables or disables this route setting configuration.
Default: enabled
Values: enabled | disabled
Example: set admin disabled
geolocation: Assigns a numeric to match on for the ip interface geolocation property. When a match occurs, the system forwards traffic to an interface with a matching geolocation. If you set the value to 0, the system uses the whole routing table when selecting an interface.
Default: 0
Example: set geolocation 10
ingress-classification-tag: Classifies (associates) incoming traffic with the configured tag. If you configure this property, you must configure the egress-classification-tag property (below) as well. The configured ingress-tag must match a configured ip routing-tag. The routing-tag identifies the groups of interfaces and routes that are available for the ingress-tag. See Tab-Based Route Selection for a complete description.
You can also configure a classification-tag through the ip interface object. If this property is configured in both places, the session-config setting takes precedence.
Note that this tag is case-sensitive.
Default: There is no default setting
Example: set ingress-classification-tag E911 egress-classification-tag: Sets an egress classification tag that is used to select the outgoing interface. That tag must then be associated with an ip > routing-tag, which controls the available egress interfaces and routes. See Tag-Based Route Selection for a complete description.
You can also configure a classification-tag through the ip interface object. If this property is configured in both places, the session-config setting takes precedence.
Note that this tag is case-sensitive.
Default: There is no default setting
Example: set egress-classification-tag E911
Enables or disables verification of the Real-Time Control Protocol (RTCP) header. The ME uses RTCP to maintain quality of service and derive diagnostic data on RTP sessions.
Enables or disables verification of the Real-Time Protocol (RTP) header. The ME uses the timing and sequence information in RTP to reassemble packets appropriately for real-time audio and video.
Configures communication between the ME and the Camiant Policy Server (3GPP Rx). The Camiant Policy Server applies business rules that determine which and when customers, tiers, and/or applications receive bandwidth priority. The ME notifies the Camiant system about media flows so that the Camiant system can then apply policy for resource allocation (or example, reserving bandwidth or setting up QoS and forwarding rules). Communication between the two system uses a 3GPP Diameter-based Rx interface.
This object sets destination Diameter servers and configures what the ME does in response to an RAR received from the Camiant (3GPP Rx) box. Using this configuration the ME is not authenticating a message, but is instead authorizing a service and reserving bandwidth for it. The Camiant box then sends the Re-Authorization Requests (RARs) when there is a network event that requires it and the ME acts according to the mapped response.
config vsp default-session-config 3GPP Rx config vsp policies session-policies policy name rule name session-config 3GPP Rx config vsp dial-plan dial-prefix entryName session-config 3GPP Rx config vsp dial-plan route name session-config 3GPP Rx config vsp dial-plan source-route name session-config 3GPP Rx config vsp session-config-pool entry name 3GPP Rx
admin: Enables or disables the Rx configuration. When enabled, the system forwards Rx requests to the configured server and takes the configured actions on RARs. When disabled, the system does not send messages to or receive them from the Camiant server.
Default: disabled
Values: enabled | disabled
Example: set admin enabled
diameter-group: Specifies a previously configured diameter-group to which the system sends Rx requests.
Default: There is no default setting
Example: set diameter-group ”vsp diameter-group 3gppGroup”
re-auth-action <cause><action>: Specifies the action the system takes on Re-Authorization Request (RAR) messages it receives from the Camiant (3GPP Rx) box. The Camiant server sends a cause, and the system takes the action mapped to that cause with this property.
Default: disconnect release-of-bearer
Values: causes:
service-info-request
charging-correlation-change
loss-of-bearer
recovery-of-bearer
release-of-bearer
establishment-of-bearer
Values: actions:
none: Take no action
re-authenticate: Attempt to reauthenticate the message by sending a new Diameter Authentication-Authorization-Request
disconnect: Disconnect the call
Example: set re-auth-action service-info-request re-authenticate
Sets parameters to ”regenerate” the SDP in order to more tightly control what is sent out by ME. This ensures that approved SDP format comes from the system on every call.
Some phones may require configuration ensuring that connection information is not specified within the media descriptor. Using the add-session-connection and remove-media-connection properties together, you can configure the ME Engine to add a session-level c-line and remove matching c-lines from the media descriptor.
The SDP c-line, which contains connection data for the session, can be found in each media description and/or at the session level. If it appears at both the session level and in the media descriptor, the media descriptor value takes precedence (for that descriptor).
config vsp default-session-config sdp-regeneration config vsp policies session-policies policy name rule name session-config sdp-regeneration config vsp dial-plan dial-prefix entryName session-config sdp-regeneration config vsp dial-plan route name session-config sdp-regeneration config vsp dial-plan source-route name session-config sdp-regeneration config vsp session-config-pool entry name sdp-regeneration
regenerate: Controls whether the system applies the settings in this object. When enabled, all settings are applied to the SDP and the system regenerates it before passing it on.
Default: enabled
Values: enabled | disabled
Example: set regenerate disabled
origin: Specifies whether the system overwrites the string that appears in the origin line (o=) of the SDP. If set to rewrite, the system changes the value of the username to CSM or the value set with the username property. In addition, it changes the value of the session-id and session-version to zero. Otherwise, it passes the name unchanged.
Default: rewrite
Values: pass | rewrite
Example: set origin-pass
username: Specifies the username to be inserted into the username field of the origin line of the SDP. This value is only applicable if the origin property is set to rewrite.
Default: There is no default setting
Example: set username joe@acme.com
session-name: Specifies whether the system overwrites the textual session name that appears in the session-name (s=) line of the SDP. If set to rewrite, the system changes the content of the session name to Oracle or the value set with the name property. Otherwise, it passes the session-name unchanged.
Default: rewrite
Values: pass | rewrite
Example: set session-name pass
name: Specifies the name to be inserted into the session-name of the SDP. This value is only applicable if the origin property is set to rewrite.
Default: There is no default setting
Example: set name ”multimedia conference”
session-info: Specifies whether to strip out or pass the textual information in the session-info (i=) line of the SDP.
Default: strip
Values: pass | strip
Example: set session-info pass
uri: Specifies whether to strip out or pass the uri (u=) line of the SDP, a pointer to additional information about the session.
Default: strip
Values: pass | strip
Example: set uri pass
e-mail-address: Specifies whether to strip out or pass the email contact information for the person responsible for the conference. This is displayed in the e= line of the SDP.
Default: strip
Values: pass | strip
Example: set e-mail-address pass
phone-number: Specifies whether to strip out or pass the telephone contact information for the person responsible for the conference. This is displayed in the p= line of the SDP.
Default: strip
Values: pass | strip
Example: set phone-number pass
bandwidth: Specifies whether to strip out or pass the proposed bandwidth to be used by the session. This is displayed in the b= line of the SDP.
Default: strip
Values: pass | strip
Example: set bandwidth pass
timing: Specifies whether to strip out or pass the start and stop times for a session. This is displayed in the t= line of the SDP. Note that some phones require a t-line for proper operation.
Default: strip
Values: pass | strip
Example: set timing pass
remove-unknown: Specifies whether to remove any unknown lines from the SDP. When enabled, all unknown (non-specification) lines are removed.
Default: enabled
Values: enabled | disabled
Example: set remove-unknown disabled
add-session-connection: Specifies whether to add session-level c-line content to the SDP if it is not already there. When enabled, the system inserts a session-level c-line (prior to the first m-line) in the SDP text message. The content for the line is derived from the first media descriptor c-line. Note that this property only adds the c-line at the session level. To remove c-lines from the media-descriptors, you must use the remove-media-connection property. See Manipulating Connection Information for more information.
Default: disabled
Values: enabled | disabled
Example: set add-session-connection enabled
remove-media-connection: Specifies whether to remove the c-line content from the SDP media descriptors. When enabled, the system removes all c-lines within media descriptors that match the session-level c-line. See Manipulating Connection Information for more information.
Default: disabled
Values: enabled | disabled
Example: set remove-media-connection enabled
add-rtpmaps: Specifies whether the system includes the rtpmap attributes for well-known CODECs (that is ”knows” about), when the rtpmap is not included by the original endpoint. Payload types under 96 must be registered with IANA as well-known CODECs. An rtpmap attribute for well-known codecs is not required in the SDP and, therefore, not included by some endpoints. However, certain endpoints have processing problems when the rtpmap is not included. When enabled, the system adds the rtpmap attributes.
Default: disabled
Values: enabled | disabled
Example: set add-rtpmaps enabled
pass-attribute: Identifies specific attribute lines to be passed through the system unchanged. Enter as many attributes as you require. Select from a predefined list or enter the attribute name. By default the system always passes certain attributes having to do with call flow (e.g., a=sendrecv) and cryptography (e.g., a=key-mgmt, a=crypto).
Default: disabled
Values: enabled | disabled
Example: set pass-attribute enabled
Sets whether or not the system re-evaluates policy for new requests on an existing session. A session, to the ME, is a SIP session between two parties, as defined by the To: and From: headers of the SIP messages.
A session is always started with a request message, such as an INVITE, SUBSCRIBE, NOTIFY, or REGISTER. Once a session is established, additional messages can be sent. For example, a request message (NOTIFY) or a MESSAGE message (containing an instant message text item) can come across an INVITE-initiated session.
Normally, the ME only processes policy (or runs all of the relevant policy rules by the message) for the first message of a session. With this setting enabled, the ME processes policy for each request message that comes across in a session.
config vsp default-session-config session-control-settings config vsp policies session-policies policy name rule name session-config session-control-settings config vsp dial-plan dial-prefix entryName session-config session-control-settings config vsp dial-plan route name session-config session-control-settings config vsp dial-plan source-route name session-config session-control-settings config vsp session-config-pool entry name session-control-settings
re-evaluate-new-requests: Specifies whether the system should process policy for all or only the first message of a session. When enabled, the system processes policy for each request message. When disabled, it processes only the first message of a session.
Default: disabled
Values: enabled | disabled
Example: set re-evaluate-new-requests enabled
refused-methods <messageType><code> <text>: Provides a method to refuse the specified SIP message type. Enter the message type with a response code and accompanying error text.
Example: set refused-methods REFER 499 ”REFER not allowed” use-lnp-for-routing: Provides a customer-specific application implementation that is not otherwise applicable.
Default: disabled
digest-realm-to-lnp-map: Provides a customer-specific application implementation that is not otherwise applicable.
Default: There is no default setting
source-lnp: Provides a customer-specific application implementation that is not otherwise applicable.
Default: There is no default setting
destination-lnp: Provides a customer-specific application implementation that is not otherwise applicable.
Default: There is no default setting
call-action-3pcc-server-entry: Sets the server that will be used when a call-control action is invoked. When a server is set, the ME processes any call control action on that server. If no server is specified, the ME determines which server to use.
Default: There is no default setting
Example: set call-action-3pcc-server-entry ”vsp enterprise 3pcc-servers internal-csta-server CSTA1”
Sets the default instruction to apply to SIP packets arriving to open a new session. The ME can either allow the session to come up, refuse it (but send a polite error response), or discard the SIP message that started the session and ignore the request.Use the SIP directive in the default session config object to apply instructions for calls in which there are no configured policies or rules; use the policy session config object to define the action for calls that match policy.
config vsp default-session-config sip-directive config vsp policies session-policies policy name rule name session-config sip-directive config vsp dial-plan dial-prefix entryName session-config sip-directive config vsp dial-plan route name session-config sip-directive config vsp dial-plan source-route name session-config sip-directive config vsp session-config-pool entry name sip-directive
directive: Sets the default SIP call directive (instruction) to apply to the SIP call session.
Default: discard
Values: allow: The system allows the packet through.
discard: The system immediately discards the packet.
refuse<resultCode><textString>: The system discards the packet but sends a response to indicate having done so. Optionally you can specify a SIP code between 400 and 699 and a text message to be appended to a refused SIP call record. If set to the refuse option, the code string is not visible.
Example: set directive refuse 401 ”Server not available”
Allows the configuration of sip-response code for calls cleared by an external H.323 GW. When a ReleaseComplete, Admission Reject, or Location Reject message is received by the ME, the ME consults an internal table to determine the appropriate SIP response code to generate when clearing the SIP side of the call. By adding a sip-response-q931-cause-map entry, you can override the internal table defaults.
config vsp session-config-pool entry sip-response-q931-cause-map config vsp default-session-config sip-response-q931-cause-map
sip-response: Define the sip-response that will be used when clearing the SIP side of the call.
Default: 0
Values: Min: 300 / Max: 699
Example: set sip-response 350
q931-cause: Select a q931-cause that helps to qualify the H.323 call-clear. If this map entry does not depend on the q931-cause value, either because there is no Q.931 present or because any Q.931 cause qualifies, choose Any.
Default: There is no default setting
Example: set q931-cause noresponse
h2250-reason: Select a h2250-reason type.
Default: none
Values: LRJ: Match an incoming LRJ message
ARJ: Match an incoming ARJ message
H.225: Match all other relevant traffic
none: H.225 reason should not be used as match criteria for this entry
Example: set h2250-reason lrj
Sets the values of SIP session timers as they are described in RFC 4028, Session Timers in the Session Initiation Protocol (SIP). These timers establish periodic refreshes of SIP sessions, through re-INVITE and UPDATE requests, to help user agents and proxies determine whether the session is still active. In other words, it provides a keepalive functionality. Use of these timers helps prevent proxies that retain state information from needlessly keeping a call active. (This may happen if the proxy, for whatever reason, does not receive a BYE from the UA.)
There are two specific timer values that are used to derive the refresh interval:
The session expiration, which defines the maximum length of a session
The minimum allowed value for the session expiration.
The ME always sends out the largest timer value that it knows of. For example, if a UAC sends a session expiration time that is lower than the ME configured timer, the ME forwards the call with its own setting. If the session expiration set by the UAC is higher, the ME uses the UAC setting. If the session expires, the configurable action determines the ME response.
The timer values that you set with this object apply to all calls that are processed through the ME device.
config vsp default-session-config sip-session-timers-settings config vsp policies session-policies policy name rule name session-config sip-session-timers-settings config vsp dial-plan dial-prefix entryName session-config sip-session-timers-settings config vsp dial-plan route name session-config sip-session-timers-settings config vsp dial-plan source-route name session-config sip-session-timers-settings config vsp session-config-pool entry name sip-session-timers-settings
admin: Specifies whether the SIP session timers configuration is in use or not. If enabled, the system adds a Session-Expires and a Min-SE header to the INVITE request.
Default: disabled
Values: enabled | disabled
Example: set admin enabled
preferred-refresher: Specifies which side of the connection performs the refreshes (send the re-INVITE and UPDATE requests). Select either UAC or UAS. If the selected device is not configured to act as a refresher, this property resets that configuration to enable refreshing (assuming the device implements the RFC).
Default: UAC
Values: UAC | UAS
Example: set preferred-refresher UAS
session-expires: Specifies the duration of the session. This is the maximum time allowed between session refresh requests before a session times out. The RFC notes that you can, but should not, set a value of less than 1800 (30 minutes), as it causes excessive messaging.
Default: 1800
Values: Min: 90 / Max: 1000000
Example: set session-expires 2700
min-se: Specifies the minimum allowed value for the session expiration interval. This lower floor is the fastest refresh rate a proxy servicing a request can require. A proxy can raise but not lower this minimum.
Default: 90
Values: Min: 90 / Max: 1000000
Example: set min-se 180 action: Specifies the action the system should take when the SIP session timers have expired.
Default: terminate
Values: terminate: The system immediately removes all internal resources dedicated to the call
disconnect: The system sends a BYE in an attempt to gracefully close the call before terminating it
nothing: The system ignores the timer expiration and keeps the session alive until it receives a BYE
Example: set action disconnect
Configures the SIP settings that the ME applies to the SIP call session. If there are no configured policies or rules to enforce on the SIP call, then the ME applies SIP settings from the default session configuration. If the call does match policy rules, the ME applies the SIP settings defined in the session-config object.
config vsp default-session-config sip-settings config vsp policies session-policies policy name rule name session-config sip-settings config vsp dial-plan dial-prefix entryName session-config sip-settings config vsp dial-plan route name session-config sip-settings config vsp dial-plan source-route name session-config sip-settings config vsp session-config-pool entry name sip-settings
mode: Sets the SIP operating mode to use with this server.
Default: auto-determine
Values: auto-determine: The system determines the mode. Usually, the system is a back-to-back user agent (B2BUA) that sends INVITE traffic to the call destination. (The system B2BUA appears as the SIP call destination, but regenerates the call to the destination SIP server.) In some cases, however, the system may act as a proxy instead.
proxy: The system is the SIP proxy that provides SIP registration, location, policy, and other services that determine the outcome of the SIP call.
Example: set mode proxy
transport: Sets the default protocol over which the SIP call session is forwarded to the destination SIP server.
Default: any
Values: any | UDP | TCP | TLS
Example: set transport udp
port: Specifies the destination port on the system for SIP traffic.
Default: auto-determine
Values: auto-determine: The system sets the SIP port.
override: Set the port number manually. Enter a port number between 1 and 65535 or leave blank to accept the default setting of 5060.
Example: set port 1212
route-hdr: Sets if and where to insert a Record-Route header in relevant SIP requests for outgoing system interfaces. This allows the system to remain in the SIP signaling path. (Route headers are used by SIP proxies only, so this behavior affects only proxied traffic.)
Default: none
Values: none: The system does not add Record-Route headers.
RouteRequest: The system inserts a Record-Route header in REQUESTs, allowing it to remain in the signaling path.
RouteRequestResponse: The system both inserts a Record-Route header in SIP REQUESTs and performs a Record-Route header fix in SIP RESPONSEs. This is needed when the SIP request and corresponding response are forwarded via different interfaces on the ME device.
Example: set route-hdrRouteRequestResponse
route-hdr-use-fqdn: Specifies where the system derives the host portion of the Record-Route header from. When enabled, the system uses a domain name in the Record-Route header. That name is either:
Configured with the route-hdr-uri-host property.
Determined from a reverse lookup on the local address that the system used to transmit the SIP message.
Default: enabled
Values: enabled | disabled
Example: set route-hdr-use-fqdn disabled
route-hdr-uri-host: Specifies the string to insert in the host portion of the Record-Route header, identifying the ME domain name. This property is for use in conjunction with the route-hdr-use-fqdn property. If you set this, be sure that the string you enter is also set as the domain name or one of the domain aliases configured in the vsp > static-stack-settings object.
Default: There is no default setting
Example: set route-hdr-uri-host acme.com
route-hdr-add-register-msg: Specifies whether to add Record-Route headers to REGISTER messages. When enabled, the system adds Record-Route headers to REGISTER messages. Enable this for compatibility with Microsoft OCS 2007. When disabled, the system does not add the headers to Record-Route headers to REGISTER messages, in compliance with the suggestions of RFC 3261, SIP: Session Initiation Protocol.
Default: disabled
Values: enabled | disabled
Example: set route-hdr-add-register-msg enabled
route-hdr-preprocess-strip: Controls whether or not the system strips the MAddr parameter off route headers prior to processing them. This property is only for use in cases where the system receives traffic from a non-RFC3261-compliant (strict-router) SIP proxy. Do not change the value unless instructed to do so by Technical Support.
Default: disabled
Values: enabled | disabled
Example: set route-hdr-preprocess-strip enabled
lcs-compatibility: Enables or disables LCS compatibility. The specific bit setting required is dependent on the desired feature, and can only be determined for your system by Technical Support. For example, if you enable the t120-anchor property of the file-transfer object, you must set this bit to 0x010032e3. Do not enable this feature unless explicitly instructed to do so.
Default: disabled
Values: enabled | disabled
Example: set lcs-compatibility enabled
in-server: Specifies the originating server in a Sametime-to-LCS interaction. This allows the system to determine the transformations it must perform to facilitate communications. For the originating server, set the server version or function. The type that you select is dependent on the server type that you are configuring.
Use this property in conjunction with the out-server property. For example, in an LCS-to-Sametime setup, the in-server type would be lcs-200x and the out-server would be sametime-31x.
Default: unknown
Example: set in-server lcs-2005
out-server: Specifies the receiving server in a Sametime-to-LCS interaction. This allows the system to determine the transformations it must perform to facilitate communications. For the receiving server, set the server version or function. The type that you select is dependent on the server type that you are configuring.
Use this property in conjunction with the in-server property. For example, in an LCS-to-Sametime setup, the in-server type would be lcs-200x and the out-server would be sametime-31x.
Default: unknown
Example: set out-server sametime-31
utilize-contact: Determines what the system should do with messages it sends out. When enabled, the system sets the address in the contact header to the ME local IP address. If disabled, the system does not form the From header from the Contact header.
Default: enabled
Values: enabled | disabled
Example: set utilize-contact disabled
add-contact-nat: When enabled, the system appends the string ”nat=true” to the contact header. If a phone is behind a firewall, and the system does not change the contact header to its own contact information, then it appends the string to the contact header.
Default: disabled
Values: enabled | disabled
Example: set add-contact-nat enabled
compress-signaling: When enabled, the system applies gzip compression to all SIP messages between the endpoints. This feature is only applicable between two ME devices.
Default: disabled
Values: enabled | disabled
Example: set compress-signaling enabled
preserve-call-id: Specifies whether the system generates a new call ID when forwarding an INVITE. Normally the system is a B2BUA, and therefore this property is disabled. As such, when it receives a call it generates a new ID for the outbound leg. If this property is enabled, the system uses the same call ID in the outbound INVITE that it received in the inbound INVITE.
Default: disabled
Values: enabled | disabled
Example: set preserve-call-id enabled
preserve-cseq: Specifies whether the system generates a new CSeq value when forwarding an INVITE. Normally the system is a B2BUA, and therefore this property is disabled. As such, when it receives a call it generates a new CSeq for the outbound leg. If this property is enabled, the system uses the same CSeq value in the outbound INVITE that it received in the inbound INVITE.
Default: disabled
Values: enabled | disabled
Example: set preserve-cseq enabled
proxy-generate-100-trying: Specifies on which type of SIP messages the system should generate and send a ”100 trying” message to the sender. The system passes the ”100 trying” when it receives it for undeclared message types.
By default, the system sends the ”100 trying” as it was received (if the system is in proxy mode). Use this mode if the message contains information that must be passed from the SIP application server to the user agent. For example, additional headers in the message may contain version information necessary for proper operation of a softphone.
When selected for a message type, the system immediately sends the initiator a ”100 trying” to indicate that it is processing the message and will forward it as soon as possible. Sending an immediate response can prevent longer SIP messages from timing out.
Default: There is no default setting
Example: set proxy-generate-100-trying register+subscribe
handle-3xx-locally: Specifies whether the system forwards responses to 3xx messages back to the UAC or resends it. If disabled, when the system receives a 3xx response for an INVITE (e.g., 300 Multiple Choices or 301 Moved Permanently), it forwards the response back to the UAC. When this option is enabled, the system does not forward the response back. Instead, it resends the INVITE message to the address specified in the contact header of the 3xx response.
Default: disabled
Values: enabled | disabled
Example: set handle-3xx-locally enabled
handle-3xx-locally-server-arbitration: Specifies whether the system uses or ignores the dial-plan > arbiter settings when forwarding a 302 Redirect message. The 302 contains the contact addresses to which the system redirects the message. If this property is enabled, the ME applies the arbiter selected for the original INVITE to the contact addresses to determine the redirect destination. If disabled, the system uses the server qvalue (preference) to select the forwarding address.
Default: disabled
Values: enabled | disabled
Example: set handle-3xx-locally-server-arbitration enabled
handle-exx-locally-lookup-original-invite: Specifies whether, in the event of a 3xx response from a redirect server, the system modifies the original INVITE it receives before doing a dial-plan lookup. If disabled, the system does a lookup on the original INVITE. When this option is enabled, the system modifies the INVITE before initiating a lookup by replacing the Request URI in the INVITE with the Contact URI found in the 3xx response. You must enable this response if you are doing a source-route dial-plan lookup. If you do not, there will be no source on which to base the lookup.
Default: disabled
Values: enabled | disabled
Example: set handle-exx-locally-lookup-original-invite enabled
session-timeout: Specifies how many seconds the system retains a session if the session did not establish successfully. For example, when authentication is required, case of authentication, the system sends a 401/407 response to the UAC to resend the request with authentication information. The session was not successful but cannot yet be terminated. This timer determines the length of time the system waits for a response.
Default: 300
Values: Min: 1 / Max: 1000000
Example: set session-timeout 500
session-duration-max: Specifies how many seconds the system maintains a session after the session has been successfully established. This property puts a timer on the session and forces a close when the timer expires. If set to 0 (the default), the session remains open until it is complete.
Default: 0 (disabled)
Values: Min: 0 / Max: 1000000
Example: set session-duration-max 18000
session-provisional-timeout: Specifies the number of seconds that the system allows the user agent server to ring before it times out the call. If forking-settings are configured, the system tries the next endstation when the timer expires, If not, the system terminates the call. If set to 0, the default, session termination due to unanswered ringing is determined by the user agent client.
Default: 0 (disabled)
Values: Min: 0 / Max: 1000000
Example: set session-provisional-timeout 30
session-authentication-timeout: Specifies the number of seconds that an INVITE session can stay in the authentication state before timing out. Because a first INVITE typically does not contain authentication information, it results in a 401 (Unauthorized) response. This timer sets the allowable time between that response and the second INVITE (which contains authentication information). If set to 0, the default, the timer is disabled and the sip-settings session-timeout property determines when the session times out.
Default: 0 (disabled)
Values: Min: 0 / Max: 1000000
Example: set session-authentication-timeout 30
outbound-local-ip: Specifies which IP address to use to reach a destination when the ME tables contain multiple addresses to that destination. Use this in conjunction with the outbound-local-port property to specify an address and port combination.
Default: There is no default setting
Example: set outbound-local-ip 10.10.10.1
max-retransmissions: Specifies a the maximum number of times the system attempts to retransmit SIP messages at the transaction level. By setting this value through the session config, you can apply different retransmission values to different message types. For example, you might want a higher number of retransmissions for a REGISTER message, and a lower number (faster response) for an INVITE. This value does not apply to OPTIONS messages. Use the settings > max-options-retransmissions property to control OPTIONS retransmissions.
Default: 1
Values: Min: 1 / Max: 32
Example: set max-retransmissions 15
outbound-local-port: Specifies which port to use to reach a destination when the ME tables contain multiple ports to that destination. Use this in conjunction with the outbound-local-ip property to specify an address and port combination.
Default: There is no default setting
Example: set outbound-local-port 3435
message-session-timeout: Specifies the number of seconds that the system keeps alive a session that was created by a MESSAGE request after the first transaction is complete.
Default: 1800
Example: set message-session-timeout 2400
udp-source-port: Specifies the destination port on the system for UDP SIP traffic.
Default: auto-determine
Values: auto-determine: The system sets the UDP port.
override: Set the port number manually. Enter a port number or leave blank to accept the default setting of 5060. Enter a port number between 1 and 65535. The default port number is 5060.
Example: set udp-source-port override 1212
add-subject-header: Adds the specified string as a Subject header when a matched dial string is present in the user portion of the REQUEST, TO, or FROM URI. The system adds the header to all requests destined for devices that have successfully registered except for REGISTER, PRACK, ACK, BYE, and CANCEL.
Default: none Values: <none | header> string <request-uri | to-uri | from-uri>
Example: set add-subject-header header internalDevice=true to-uri
strip-route-header: Specifies whether the system should strip out the route header and ignore the information it contains on both inbound and outbound traffic. The route header is a field of the SIP header which is not commonly used, but when it is, the system gives it highest priority when forwarding calls.
By default (when disabled), the system uses the information in the route header to forward packets. Set this property to enabled to ensure that manual configuration of SIP routing takes precedence.
Default: disabled
Values: enabled | disabled
Example: set strip-route-header enabled
strip-via-headers: Specifies whether to remove all but the specified number of Via headers when forwarding a request. When enabled, the system strips all but the specified number of Via headers from the request as the request passes through the box. For example, if set to 1, the system strips all Via headers except the bottom (furthest). The system always inserts itself as the top Via header, regardless of the setting. It then restores the Via headers on the response. When disabled, the default, the system retains all Via headers. If enabled, the default number of headers kept is 0 (strip all).
Default: disabled Values: enabled | disabled integer
Example: set strip-via-headers enabled 3
sticky-via: Specifies whether to replace the port number in a VIA header. This property is only for use in cases where a unique port is assigned at connection and that port should be used in the VIA header. When this property is set to 0, the system leaves the port at the value that was received. When set to 1, the system replaces the value in the top VIA header. When set to 2, the system replaces the value in the next-to-last VIA header. Only use this property if instructed to do so by Technical Support.
Default: 0
Values: 0 | 1 | 2
Example: set sticky-via 1
ignore-provisional-tag: Specifies whether the system updates its internal SIP session using the TO header tag present in the provisional response sent by an intermediate server. If enabled, the system ignores the provisional response and updates the SIP session with the TO header tag found in the SIP final response. When disabled, the system updates the SIP session with the TO header from the provisional response.
Default: enabled
Values: enabled | disabled
Example: set ignore-provisional-tag disabled
forward-provisional-ack: Specifies which points along the path respond to a provisional ACK. When enabled, the mid-stations along the path do not respond to the PRACK message. Only the endpoints respond (end-to-end). When set to disabled, the default, the each hop along the path responds.
Default: disabled
Values: enabled | disabled
Example: set forward provisional-ack enabled
terminate-transaction-on-bye: Specifies system behavior when it receives a BYE message. If enabled, the ME device terminates any outstanding transactions on that leg when it receives a BYE message. The system applies the termination to a particular call leg for a particular session. When disabled, the system does not terminate outstanding transactions when it receives a BYE. Instead, it only terminates them when either it receives the final response belonging to the transaction or the transaction times out.
Default: enabled
Values: enabled | disabled
Example: set terminate-transaction-on-bye disabled
to-header-follows-contact-header: Specifies that the system should move the URI and header parameters from the Contact header to the To header. This property is used for a specific phone switching application only.
Default: disabled
Values: enabled | disabled
Example: set to-header-follows-contact-header enabled
inleg-tos: Determines the TOS value setting for the in-leg of the session. The TOS value determines the quality of service that the call receives. If set to preserve, the system uses the TOS value in the first received message of the session (normally a REGISTER or INVITE). If set to overwrite, the system marks the TOS field of all packets it sends out on the inleg with the value you specify. Enter a number that represents the 8-bit Differentiated Services (DS) field of the IP packet in decimal format, such as 26 for 00011010 or104 for 01101000. This value can be of use to upstream devices.
Default: preserve; if you select overwrite, the default value is 0 Values: preserve | overwrite value
Example: set inleg-tos overwrite 22
outleg-tos: Determines the TOS value setting for the out-leg of the session. The TOS value determines the quality of service that the call receives. If set to preserve, the system uses the TOS value in the first received message of the session (normally a REGISTER or INVITE). If set to overwrite, the system marks the TOS field of all packets it sends out on the outleg with the value you specify. Enter a number that represents 8-bit Differentiated Services (DS) field of the IP packet in decimal format, such as 26 for 00011010 or 104 for 01101000. This value can be of use to upstream devices.
Default: preserve; if you select overwrite, the default value is 0 Values: preserve | overwrite value
Example: set outleg-tos overwrite 22
generate-final-response: Sets whether the system generates a final response when it cannot forward an OPTIONS request or does not receive a response. When enabled, the system sends a 408 Request Timeout back to a UAC if it does not receive a response to an OPTIONS message from the UAS. When disabled, it does not generate a 408.
Default: enabled
Values: enabled | disabled
Example: set generate-final-response disabled
b2bua-generate-100-trying: Sets whether the system sends a 100 Trying message back to the caller before forwarding an INVITE. When enabled, the message is sent.
Default: enabled
Values: enabled | disabled
Example: set b2bua-generate-100-trying disabled
auto-accept-reinvite-with-no-sdp-on-in-leg: Specifies whether the system responds with a 200 OK or forwards a REINVITE. When a call is first established, the system receives an INVITE on the in-leg and forward it out the out-leg. Once the call has been established, either side may send a REINVITE. If this property is enabled and the call is received on the in-leg, the system responds with a 200 OK. If disabled, the system forwards the REINVITE to the out-leg.
Default: disabled
Values: enabled | disabled
Example: set auto-accept-reinvite-with-no-sdp-on-in-leg enabled
auto-accept-reinvite-with-no-sdp-out-leg: Specifies whether the system responds with a 200 OK or forwards a REINVITE. When a call is first established, the system receives an INVITE on the in-leg and forward it out the out-leg. Once the call has been established, either side may send a REINVITE. If this property is enabled and the call is received on the out-leg, the system responds with a 200 OK. If disabled, the system forwards the REINVITE to the in-leg.
Default: disabled
Values: enabled | disabled
Example: set auto-accept-reinvite-with-no-sdp-out-leg enabled
strip-authint-qop: Determines whether the system modifies the Quality of Protection (QoP) parameter. The QoP parameter defines the type of authentication the server requires, either auth or auth-int. Auth verifies the sender using a shared secret; auth-int verifies both the sender and the integrity of the message (as defined in RFC 2617, HTTP Authentication: Basic and Digest Access Authentication). When enabled, if the QoP parameter of a 401 or 407 challenge response offers both auth and auth-int, the system removes the auth-int option and forwards the message with just auth required. If disabled, the QoP parameter remains unchanged.
Default: disabled
Values: enabled | disabled
Example: set strip-autnint-qop enabled
ignore-cancel-branch: Specifies whether to use the branch value to cancel a transaction. By default (disabled) the system uses the branch value (a string that identifies the transaction) in the top VIA header. When enabled, the system uses the FROM and TO tags in the call ID to identify and cancel the transaction. Use this property only if a device sends the wrong branch value.
Default: disabled
Values: enabled | disabled
Example: set ignore-cancel-branch enabled
symmetric-signaling: Sets whether the system uses the VIA or CONTACT header to send messages for a call. An incorrectly functioning NAT device may only partially change a CONTACT or VIA header, causing the system to be unaware that the sending endpoint is behind a NAT. When enabled, this property ensures that the system always sends responses and future requests for that call to the IP address and port number from which the request originated. Enable this property only if instructed to do so by Technical Support.
Default: disabled
Values: enabled | disabled
Example: set symmetric-signaling enabled
sips-uri-scheme-setting: Controls the SIP scheme the system uses in the URI of outbound calls. For example, when one endpoint is using TLS and another UDP, the SIP scheme is inconsistent. The default setting, auto, allows the system to determine the scheme. To force the setting, select either secure (for sips) or not-secure (for sip).
Default: auto
Values: auto | secure | not-secure
Example: set sips-uri-scheme-setting not-secure
redirect-preserve-session-config: Specifies where the system derives the session configuration from when a WSDL-generated INVITE results in a 3xx response. When enabled, the system applies the WSDL request's session configuration to the redirect message (the session configuration within the WSDL request is maintained throughout the session). When disabled, the default, the system uses the session configuration associated with the original INVITE for the redirect message as well, rather than the WSDL-based session configuration.
Default: disabled
Values: enabled | disabled
Example: set redirect-preserve-session-config enabled
max-forwards: Specifies the value to insert into the Max-Forward header of the SIP message. This value to determine how many future hops the message is allowed, and is decremented by 1 at each hop. If a message arrives with a Max-Forwards value higher than the value set with this property, the system resets the value to this configuration setting. If the value that arrives is equal to or lower than this setting, the system decrements that value by one.
Default: 70
Example: set max-forwards 50
enum-fail-response: Specifies the customized response code and string that the system sends as a result of a failed ENUM lookup. When set to ignore, the system ignores the failed lookup and continues to try and transmit the message. (The system may be able to resolve the address via a configured dial plan, for example.) If it is not able to ultimately resolve the TO header, the system sends a 404 Not Found message back to the sender. When set to reject, the system rejects the incoming message immediately and sends the configured code and string to the sender. You may use this, for example, if the originating phone/gateway requires a custom cleardown (SIP response) when ENUM lookup fails.
Default: ignore Values: ignore | reject [result-code][result-string]
Example: set enum-fail-response reject 999 ”ENUM failure”
dns-fail-response: Sets the response code that the system sends to an endpoint when a call is received, the dns-client-settings > routing-last-resort-dns property is applied, and the lookup fails.
Default: 404
Example: set dns-fail-response 400
dns-fail-response-string: Sets the response string that the system sends to an endpoint when a call is received, the dns-client-settings > routing-last-resort-dns property is applied, and the lookup fails.
Default: Not Found
Example: set dns-fail-response-string ”Bad Request
update-moc-via-csta: Specifies whether to send user state information relevant to the Microsoft Office Communicator (MOC) client. If enabled, the system notifies the MOC of the change in call state via CSTA when a call is connected or terminated. This provides MOCs that are logged in with remote-call-control enabled with correct presence information (in-a-call or available).
Default: disabled
Values: enabled | disabled
Example: set update-moc-via-csta enabled
supported-inleg: Adds a new Supported header or overwrites the existing header on the inbound leg of the SIP message with the specified text string. Use this property, for example, to support Provisional Response Acknowledgement (RFC 3262) PRACK insertion.
Default: There is no default setting
Example: set supported-inleg 100rel, timer
supported-outleg: Adds a new Supported header or overwrites the existing header on the outbound leg of the SIP message with the specified text string. Use this property, for example, to support Provisional Response Acknowledgement (RFC 3262) PRACK insertion.
Default: There is no default setting
Example: set supported-outleg ”Supported: 100rel”
persistent-destination: Specifies whether to send a message to a new remote address based on a new remote contact. When set to true, the system ignores the Contact URI and forwards messages to their original destination address. When set to false, the system uses the Contact URI in each response to determine the destination of the next message.
Default: true
Values: true | false
Example: set persistent-destination false
loop-detection-threshold: Specifies how many times a message can be processed by the system before it is considered to be in a loop. If this threshold is reached, the system rejects the call and sends a 482 (Loop Detected) response code back to the caller.
Default: 2
Values: Min: 1 / Max: 256
Example: set loop-detection-threshold 4
sip-signaling-encryption: Specifies whether a call coming in over a TLS connection is allowed or dropped.
Default: unspecified
Values: unspecified: Any transport protocol is allowed; the system continues processing regardless of the connection type.
prohibited: TLS is not allowed. If a call comes in over a TLS connection, the system drops the call.
required: Any transport protocol is allowed; the system continues processing regardless of the connection type.
Example: set sip-signaling-encryption required
share-transport-connection: Specifies when the ME should reuse existing TCP or TLS connections. The setting specifies the criteria for matching between the existing connection and the current transaction (request or response). The ME reuses the connection. If there is no match, the system opens a new connection.
Default: remote-addr
Values: remote-ip: The remote IP address
remote-addr: The remote IP address and phone number
no-local-port: The local IP address, the remote IP address, and the remote port
all: The remote port and IP address and the local port and IP address
Example: set share-transport-connection all
propagate-RR-headers: Secondary property. Modifies headers in a B2B configuration. When enabled, the system propagates Record Route and Via headers from received to transmitted SIP messages. When disabled, the system creates these headers for each transmitted message. Do not modify this setting unless instructed to do so by Technical Support.
Default: disabled
Values: enabled | disabled
Example: set propagate-RR-header enabled
ignore-via-port: Secondary property. Specifies whether to ignore the port reported in the top-level Via header. Typically, when the system matches a request to an existing transaction (request/response), it checks to make sure that, among other fields, the port is the same in both. This property should be enabled if there is a device along the path that incorrectly changes the Via header port. By default, this property is disabled (port numbers are checked).
Default: disabled
Values: enabled | disabled
Example: set ignore-via-port enabled
ignore-route-header: Secondary property. Specifies whether the ME uses the Route header to forward messages. By default (disabled), the system does use the Route header. When enabled, the ME message forwarding logic does not use the Route header and instead uses other options, as set in the session configuration (Request URI, Contact URI, DNS, etc.).
Default: disabled
Values: enabled | disabled
Example: set ignore-route-header enabled
make-provisional-resp-reliable: Secondary property. Specifies a single provisional response code to which the system will add ”Required: 100rel” and ”RSeq: nnn” before forwarding. These headers are only added, however, if the initial INVITE indicates support for 100rel. Note that the forward-provisional-ack property should be disabled when using this so that the PRACK from the UAC wont be forwarded to the UAS. If you are using third-party call control, the use-183-for-ringing-with-sdp converts a 180 response to a 183. In this case, set this property to 183 instead of 180.
Default: 0
Example: set make-provisional-resp-reliable 3
allow-redirect: When enabled, the ME is able to redirect incoming calls to other servers.
Default: enabled
Values: enabled | disabled
Example: set allow-redirect disabled
strip-received-from-top-via: When enabled, the ME strips off the ”Received” and ”Rport” fields in the top header before it sends a 200OK response.
Default: disabled
Values: enabled | disabled
Example: set strip-received-from-top-via enabled
last-resort-request-uri: If both the dial plan and location cache look ups fail, when this parameter is enabled, the ME attempts to route the call using the Request-URI of the incoming INVITE. If you want to limit routing to dial plans and the location cache, set this parameter to disabled.
Default: enabled
Values: enabled | disabled
Example: set last-resort-request-uri disabled
preserve-session-config-on-3xx: Apply the same session configuration that was used for the initial INVITE when sending the INVITE for a 302. When disabled, the AA-SBC removes the dial plan and server session configurations from the merged configuration.
Default: disabled
Values: enabled | disabled
Example: set preserve-session-config-on-3xx
Configures call control, allowing the ME or a CSTA client to control (become the third party) in a call. Specifically, this object controls the WAV files that the ME should play and the external status events reported to an external server for calls created by the ME device. The third-party call controller (3PCC) functionality is used, for example, to enable the interworking of a uaCSTA with the Broadworks Open Client Interface. The ME converts between the two call control protocols. Phone control can be integrated, for example, into Microsoft Office applications using the Phone Controls interface. This object can also be enabled in certain situations involving LCS/Sametime interworking and other advanced the ME applications. In all cases, it should only be enabled at the direction of Technical Support.
When the ME functions as a 3PCC device by initiating communications to each endpoint in the session, this object configures the specific WAV file(s) to play in response to the state of the call destination. Specifically, when the ME receives an instruction from the CSTA client to establish a call, it first makes a call to the originator of the call and then to the destination. The destination responds with call progress information. If that information indicates that the phone is ringing, and the ringback-file property is configured, the ME plays the specified file. If the phone is busy or set to appear so, the ME plays any configured busy-file recording.
Note:
You must set the admin property of this object to enabled if you are implementing a Sametime-to-LCS federation. If you are running Sametime-to-Sametime or LCS-to-LCS, the admin property must be disabled.When the pre-call-announcement property is set, the ME plays a WAV file for the caller prior to the call connecting. If you set this property and want to ensure the pre-call announcement is played in the event that the master box fails over to a backup box, you must copy files to the backup boxes using the file mirror service. To do so:
Enable the file-mirror master service and configure a file-mirror-directory. List all boxes as possible hosts with the host-box property. For example:
config file-mirror> set admin enabled config file-mirror> set file-mirror-directory /cxc_common/mirror config file-mirror> set host-box cluster\box 1 config file-mirror> set host-box cluster\box 2 config file-mirror> set host-box cluster\box 3
Upload the pre-call announcement WAV file to the configured mirror directory.
Execute the file-mirror-service make-available action. For example:
NNOS-E> file-mirror-service make-available /cxc_common/mirror/announcement.wav
By executing the action after uploading files to the file mirror directory, these files will be available to master and drone in the event of failover.
config vsp default-session-config third-party-call-control config vsp policies session-policies policy name rule name session-config third-party-call-control config vsp dial-plan dial-prefix entryName session-config third-party-call-control config vsp dial-plan route name session-config third-party-call-control config vsp dial-plan source-route name session-config third-party-call-control config vsp session-config-pool entry name third-party-call-control
admin: Enables or disables the ability to control externally originated calls. (If the system is the call originator, this setting is ignored.)
Default: disabled
Values: enabled | disabled
Example: set admin enabled
status-events: Specifies when to generate status events if the system is functioning as a third-party call controller (3PCC), initiating communications to each endpoint in the session. Specifically, when the system receives an instruction from the CSTA client to establish a call, it first makes a call to the originator of the call and then to the destination.
Default: both
Values: originate: Contact established with the call originator
answer: Contact established with the call destination
neither: No status events are sent
both: Contact established with the call originator and destination
Example: set status-events originate
handle-refer-locally: Specifies whether the system forwards REFER messages or whether it consumes the message and generates an INVITE for the call. When an end user wants to transfer a call, the system receives a REFER message. When this property is set to enabled (the default), the system terminates the message, generates an INVITE, and makes a call on behalf of that call transfer. When set to disabled, the system forwards the call to the upstream server for handling.
Default: enabled
Values: enabled | disabled
Example: set handle-refer-locally disabled
refer-maintain-identity: Specifies how the system handles the TO and FROM URIs when processing a REFER request. If set to true, the system keeps the original TO and FROM components of the URI instead of replacing them with the REFER destination. When set to false, the default, they are replaced.
For example, suppose a third-party call is initiated from A to B, and B subsequently sends a REFER to transfer the call. If this property is set to true, the call appears to A as if it were still connected to B. If set to false, the call appears to A as if it were connected to C.
Default: false
Values: true | false
Example: set refer-maintain-identity true
ringback-file: Specifies the WAV file that the system plays while the remote endpoint is ringing. If the system is the call originator, and this property is configured, the system plays the file regardless of the admin setting. If the call is externally originated, the ringback is generated locally by the originating phone.
Default: There is no default setting
Example: set ringback-file /cxc/cxc_common/holdRigning.wav
busy-file: Specifies the WAV file that the system plays when the remote endpoint is set to do-not-disturb or is busy. If the system is the call originator, and this property is configured, the system plays the file regardless of the admin setting. If the call is externally originated, the busy signal is generated locally by the originating phone.
Default: There is no default setting
Example: set busy-file /cxc/cxc_common/busy1.wav
pre-call-announcement: Specifies the WAV file that the system plays for the caller when this property is set. The callee does not hear the file (the file is played before the connection to the destination is attempted). In contrast, use the introduction property of the media object to play an recording for both sides.
Default: There is no default setting
Example: set pre-call-announcement cxc/'cxc_common/youhavewon.wav
terminate-after-pre-call-announcement: Specifies whether to end a call once the pre-call announcement has been played. When enabled, the system plays the file specified with the pre-call-announcement property and then terminates the call.
Default: disabled
Values: enabled | disabled
Example: set terminate-after-pre-call-announcement enabled
handle-replaces-locally: Specifies whether the system forwards INVITEs with a Replaces header or whether it consumes the message and generates a new INVITE for the call. When an endpoint transfers a call, the ME receives a REFER message. When this property is enabled, the system terminates the message and completes the replacement locally. When disabled, the system forwards the call to the upstream server for handling.
Default: disabled
Values: enabled | disabled
Example: set handle-replaces-locally enabled delayed-ack: Specifies when the system sends an ACK in response to an originating call leg. When enabled, the system does not send an ACK to the originating call leg until it receives a 200/OK from the answering call leg. When disabled, the system sends an ACK immediately, prior to receiving a response from the destination.
Default: disabled
Values: enabled | disabled
Example: set delayed-ack enabled
include-reason-in-bye: Specifies whether to include the optional Reason header in a BYE request. Enable this to resolve the Heterogeneous Error Response Forking Problem (HERFP), as stated in RFC 3326. The Reason field helps in properly updating requests when proxy servers forward requests to multiple contacts associated with a call. When enabled, the default, the Reason header is included.
Default: enabled
Values: enabled | disabled
Example: set include-reason-in-bye disabled
always-apply-req-uri-spec: Specifies when to include the settings of the request-uri-specification. When enabled, the system applies the settings to the initial INVITE as well as any re-INVITEs. When disabled, the system only applies the settings on the initial INVITE.
Default: enabled
Values: enabled | disabled
Example: set always-apply-req-uri-spec disabled media-shuffle: Specifies whether to generate SDP locally in H.323-to-SIP calls when the H.323 endpoint is not in fast-start mode. (Fast-start mode is the option of having the H.323 endpoint provide media information in the initial call setup.) This property is used for injecting SDP into INVITEs for the SIP Delayed Offer to Early Offer conversion process.
In a non-fast-start case, the system has no media information when it sends out an INVITE. When this property is enabled, the system generates some SDP initially. When it receives the real media information from the H.323 side, the system then reinvites the SIP endpoint with the correct information. When media-shuffle is disabled, the system sends the INVITE with no SDP.
Default: enabled
Values: enabled | disabled
Example: set media-shuffle disabled
park-incoming-calls: Allows establishment of an inbound 3PCC session. When enabled, any incoming call is answered by the system (similar to the results of the call-control park action). Use this, for example, in conjunction with the pre-call-announcement property.
Default: disabled
Values: enabled | disabled
Example: set park-incoming-calls enabled
parked-call-greeting: Specifies the WAV file that the system plays for the caller when the park-incoming-calls property is set to enabled. The callee does not hear the file (the file is played before the connection to the destination is attempted). In contrast, use the introduction property of the media object to play a recording for both sides.
Default: There is no default setting
Example: set parked-call-greeting /cxc/cxc_common/park.wav terminate-after-greeting: Specifies whether the ME terminates a call after playing the file specified with the parked-call-greeting property. The park-incoming-calls property must be set to enabled for this property to apply.
Default: disabled
Values: enabled | disabled
Example: set terminate-after-greeting enabled
terminate-update-locally: Specifies whether the system responds locally to UPDATE messages. When this is enabled, if an endpoint sends an UPDATE message, the system responds to the update with a 200 OK and the SDP from the remote endpoint. If disabled, the system forwards the UPDATE to the remote endpoint.
Default: disabled
Values: enabled | disabled
Example: set terminate-update-locally enabled
terminate-reinvite-locally: Specifies whether the system responds locally to REINVITE messages. When this is enabled, if an endpoint sends a REINVITE message, the system responds to the reinvite with a 200 OK and the SDP from the remote endpoint. If disabled, the system forwards the REINVITE to the remote endpoint.
Default: disabled
Values: enabled | disabled
Example: set terminate-reinvite-locally forking-early-media-inhibit: Specifies whether to strip SDP from a provisional response. Use this in some cases when using sequential or parallel forking to prevent an endpoint from being swamped by provisional response content. The saved SDP is later inserted into the 200 OK (unless more current SDP is available in the 200 OK), allowing the calling phone to receive just one SDP.
Default: disabled
Values: enabled | disabled
Example: set forking-early-media-inhibit enabled
forward-unresolved-replaces: Secondary property. Specifies whether to forward a 481(Call Leg Not Found) message when the system cannot find the call leg information in the Replaces header. In an attended transfer, A calls B, and then B calls C, makes the connection, and completes the transfer. To do so, B sends a REFER to C. Included in that REFER is a Replaces header, with information identifying the call-leg from A to B (which will be replaced). As a B2B, the system must convert the identifying information in the call leg on the B side to that which represents the call-leg on the A side. When this property is disabled, if the system cannot find the call-leg in the Replaces header it responds with a 481 to the system receiving the REFER. If enabled, the system forwards the REFER to C without the conversion described.
Default: disabled
Values: enabled | disabled
Example: set forward-unresolved-replaces enabled extract-refer-to-header: Secondary property. Specifies whether to use the fields of the REFERTO header. When enabled, the ME extracts the header specifications from the REFERTO header and includes them in the resulting INVITE.
Default: disabled
Values: enabled | disabled
Example: set extract-refer-to-header enabled
reinvite-preserve-media: Secondary property. Specifies whether to re-use SDP. When set to enabled, in cases where an REINVITE is received and there is no SDP, the ME uses the previous SDP.
Default: disabled
Values: enabled | disabled
Example: set reinvite-preserve-media enabled
media-forward: Secondary property. Specifies whether the ME can respond to forwarded requests from the NICE media server. When this property is enabled, the system treats all INVITEs as requests from the NICE media server. (If the INVITE is not from the NICE media server, the system will reject the request after scanning and not finding NICE-specific content.)
Default: disabled
Values: enabled | disabled
Example: set media-forward enabled
track-to-user: Secondary property. Allows the NICE media server to request forwarding sessions based on the user listed in the TO URI.
Default: disabled
Values: enabled | disabled
Example: set track-to-user enabled force-retrieve-on-delayed-offer-while-held: Secondary property. Specifies system behavior when an INVITE with no SDP is received for a call on hold. When enabled, if an INVITE with no SDP is received, it is treated as a receive request and the call is taken off hold. When disabled, call status does not change from that event.
Default: disabled
Values: enabled | disabled
Example: set force-retrieve-on-delayed-offer-while-held enabled
reinvite-delayed-offer-wait-on-ack: Secondary property. Specifies how the ME responds to a REINVITE with no SDP. When enabled, the ME postpones re-inviting the remote end (accepting the REINVITE locally) until receiving the SDP contained in the ACK. The default setting is disabled.
Default: disabled
Values: enabled | disabled
Example: set reinvite-delayed-offer-wait-on-ack enabled
use-183-for-ringing-with-sdp: Secondary property. Specifies the SIP code that the ME forwards with messages it receives that contain early media. If this property is disabled, the system forwards the message with a 180 (Ringing) code. If enabled, the system changes the code to 183 (Session Progress) when forwarding the message.
Default: disabled
Values: enabled | disabled
Example: set use-183-for-ringing-with-sdp enabled strip-require-100-rel: Secondary property. Specifies whether the ”Require: 100rel” line is stripped from responses. When enabled the line is removed from forwarded responses.
Default: disabled
Values: enabled | disabled
Example: set strip-require-100-rel enabled forward-all-parallel-provisional-responses: Secondary property. Sets how provisional responses are handled when parallel forking is enabled. Parallel forking results in a single incoming INVITE generating multiple simultaneous outgoing INVITEs, which can cause issues regarding which audio stream to apply if multiple provisional responses are returned. When this property is enabled, if a provisional response is received during a parallel forking attempt, every 18x response received prior to one of the outgoing calls completing successfully is forwarded to the originating agent. If this property is disabled, provisional responses are only forwarded when received from the first server on the list of destinations being attempted.
Default: disabled
Values: enabled | disabled
Example: set forward-all-parallel-provisional-responses enabled
include-id-in-refer-notify: Secondary property. Specifies whether, when a NOTIFY is sent reporting the final status of a REFER call, the ME includes an ID field in the NOTIFY. When enabled, the ME includes the ID.
Default: disabled
Values: enabled | disabled
Example: set include-id-in-refer-notify enabled notify-dtmf-event-if-allowed: Secondary property. Sets whether the system sends a NOTIFY event in response to receiving DTMF. To use this property, the remote user agent must allow KPML or telephone events. When the ME receives an Allow-Events header, and this property is enabled, upon receiving DTMF the system sends a NOTIFY event with the RFC 2833 representation of the received DTMF.
Default: disabled
Values: enabled | disabled
Example: set notify-dtmf-event-if-allowed enabled
terminate-hold-retrieve-locally: When this property is enabled, if a re-INVITE is received with an SDP that indicates it is either a hold or retrieve request, the ME accepts the re-INVITE locally with the SDP acknowledging the hold or retrieve and the message is not forwarded. When this property is disabled, re-INVITE messages with an SDP that indicates hold or retrieve receive no special treatment.
Default: disabled
Values: enabled | disabled
Example: set terminate-hold-retrieve-locally enabled
reinvite-originator: When enabled, the ME reinvites the original UAC after the call is initially set up.
Default: disabled
Values: enabled | disabled
Example: set reinvite-originator enabled
skip-shuffle-complete-if-anchored: When enabled, no reinvite is sent forwarding the SDP contained in the ACK for calls with anchored media.
Default: disabled
Values: enabled | disabled
Example: set skip-shuffle-complete-if-anchored enabled forward-302-division-header: Secondary property. When enabled, if a 302 Redirected response with a Diversion: header is received by the ME, the Diversion: header is forwarded in the response.
Default: enabled
Values: enabled | disabled
Example: set forward-302-division-header disabled strip-route-headers: Secondary property. Specifies which route headers back-to-back user-agents strip.
Default: all
Values: none
all
top
Example: set strip-route-headers none
inhibit-shuffle-update: Controls whether a re-INVITE message is generated and sent to the destination server to update the SDP (as some user agents initiate SIP calls using an INVITE without an SDP.)
Normally, when the media-shuffle property is enabled, the SDP is generated locally for the outgoing call, and the answer SDP from the destination will be returned to the originator as the offer SDP. After the originator responds with the answer SDP, a reinvite will be generated to the destination to update the SDP.
When inhibit-shuffle-update is set to enabled, this re-INVITE will not be sent to update the SDP. The inhibit-shuffle-update property setting is only valid when media-anchoring is enabled. If inhibit-shuffle-update is enabled without media anchoring enabled, audio problems will result.
Default: disabled
Values: enabled | disabled
Example: set inhibit-shuffle-update enabled
refer-notify-100-trying: Controls the behavior of the ME when a REFER message is received and the referrer disconnects before the resulting transfer has completed.
Example: set refer-notify-100-trying attempt1 forward-302-diversion-header: When enabled, if a 302 Redirected response with a Diversion: header is received by the ME, the Diversion: header is forwarded in the response.
Default: enabled
Values: enabled | disabled
Example: set forward-302-diversion-header disabled
media-forward-reference-direction: Identifies which leg of a call is to the call-center PBX, mapping the Rx and Tx streams to match the NICE equipment Rx and Tx streams.
Default: out-leg
Values: in-leg | out-leg
Example: set media-forward-reference-direction in-leg
inhibit-100-trying-for-reinvite: When enabled, the ME does not send out a 100 Trying when it receives a re-INVITE. When disabled, the ME does send out a 100 Trying in response to a re-INVITE.
Default: enabled
Values: enabled | disabled
Example: set inhibit-100-trying-for-reinvite disabled
allow-lcr-for-refer: When running the route server under third-party-call-control, information may need to be obtained off of the ME, causing a delay. When this parameter is enabled, the ME can avoid potential problems caused by this delay.
Default: disabled
Values: enabled | disabled
Example: set allow-lcr-for-refer enabled transfer-file: Select the file of the media to be played while a blind transfer is taking place.
Default: There is no default setting
Example: set transfer-file data1
notify-dtmf-when complete: Secondary property. Specifies whether the ME forwards the DTMF notification via the Notify request sent at the beginning of the DTMF tone or the Notify request sent after the DTMF tone.
Default: enabled
Values: enabled | disabled
Example: set notify-dtmf-when-complete disabled
inhibit-provisional-response-after-prack: Secondary property. When enabled, the ME does not send 18x messages after receiving a Prack. This prevents problems when invalid codecs are presented in the 18x's SDP after valid codecs have already been sent.
Default: disabled
Values: enabled | disabled
Example: set inhibit-provisional-response-after-prack enabled
call-control-events-version: When custom-event-fields object on the ME is configured, when this property is set to custom, it enables the ME to add custom information in call control events (for example, callCreated, callConnected, and callTerminated).
Default: legacy
Values: legacy: The events generated follow the legacy format.
custom: The events generated have new custom fields.
Example: set call-control-events-version custom
terminate-message-locally: Indicates whether the ME forwards a received SIP MESSAGE after responding with a 200 OK.
Default: disabled
Values: enabled | disabled
Example: set terminate-message-locally enabled
inhibit-outgoing-reinvites: Specifies whether the ME responds locally to REINVITE messages. When enabled, the ME response to the REINVITE with a 200 OK and SDP from the remote endpoint before forwarding the message. When disabled, the ME forwards the REINVITE to the remote endpoint.
Default: disabled
Values: enabled | disabled
Example: set inhibit-outgoing-reinvites enabled
forward-provisional-media: Indicates whether the ME plays a ringback-file. When enabled, the ME does not play the ringback-file when establishing a third-party initiated call and the out-leg 18X contains SDP.
Default: disabled
Values: enabled | disabled
Example: set forward-provisional-media enabled
disconnect-time: Secondary property. Sets the number of seconds from the time a session is established until it is terminated. Reinvites and updates do not refresh this time.
Default: 0
Values: Min: 0 / Max: 10000000
Example: set disconnect-time 100000
inhibit-anchored-update: Specifies whether a re-INVITE message is generated and sent to the destination server to update the SDP for a third-party initiated call.
Note:
This property is valid only when the session-config > media > anchor property is set to enabled.
Default: disabled
Values: enabled | disabled
Specifies what derives the content of the fields of the TO URI when the ME transmits a message. For example, if the user property is set to to-uri, the ME replaces the user field of the TO URI with data from the user field of the incoming TO URI. If set to omit, the user field is left blank. Or, you can enter any string that you want placed in the user field.
In some cases, it is necessary to change a portion of the URI (outbound only) so that the next-hop server can accept the SIP message when it arrives. For example, a server may require a part of the URI to be in a specific format. The ME allows you to modify the portions of the REQUEST, TO, and/or FROM URI so that the header matches any necessary requirements.
When you select to alter the URI, you can set the ME to replace the specified field with one of the following. Properties have some or all of the same selection options. These options apply to the to-uri-specification, from-uri-specification, request-uri-specification, and inbound-controls objects. The following defines the common options, where the ME device:
request-uri: Derives values from the incoming REQUEST URI
to-uri: Derives values from the incoming TO URI
from-uri: Derives values from the incoming FROM URI
omit: Leaves the field blank
next-hop: Derives values from the IP address of the next-hop server (not applicable to the port field)
local: Derives values from the IP address of the ME device (not applicable to the transport field)
omit-phone-context: Does not replace the field, but removes the phone context portion, if applicable (user property only)
next-hop-domain: Derives values from the IP address of the next-hop server or phone. If a server, the next-hop is learned from the peer property of the server; if a phone, it is the IP address of the phone (host property only)
next-hop-ip: Derives values from the next-hop IP address, which is specified in the peer property of the same object (host property only)
local-ip: Derives values from the IP address from the interface the packet goes out on (host property only)
directory directoryReference: Derives values from a user alias. When selected, the ME looks for all aliases associated with the user listed in the To, Request, and From fields of the URI. The ME then uses the alias associated with the referenced directory. (host property only)
string: Writes the specified string to the field (not applicable to the transport field)
none or same-uri: The URI is not modified
For example, it is not uncommon for a carrier to change a user ANI (automatic number identification) to the modified number used by the DNIS (dialed number identification service). For the ME to correctly forward the call, it must put the user ANI back into the From header. It can do this if you configure the host property to a referenced directory that can identify the user DNIS alias.
config vsp default-session-config to-uri-specification config vsp policies session-policies policy name rule name session-config to-uri-specification config vsp dial-plan dial-prefix entryName session-config to-uri-specification config vsp dial-plan route name session-config to-uri-specification config vsp dial-plan source-route name session-config to-uri-specification config vsp session-config-pool entry name to-uri-specification
user: Specifies how to derive the value of the user field of the TO URI.
Default: to-uri Values: request-uri | to-uri | from-uri | omit | next-hop | local | omit-phone-context | string
Example: set user request-uri
host: Specifies how to derive the value of the host field of the TO URI.
Default: to-uri Values: request-uri | to-uri | from-uri | omit | next-hop | next-hop-domain | local-ip | string
Example: set host omit
port: Specifies how to derive the value of the port field of the TO URI.
Default: to-uri Values: request-uri | to-uri | from-uri | omit | string
Example: set port omit
display: Specifies how to derive the value of the display field of the TO URI.
Default: to-uri Values: request-uri | to-uri | from-uri | omit | next-hop | string
Example: set display next-hop
transport: Specifies the value of the transport field of the FROM URI. In addition to using the value from other fields of the incoming URI, you can set the transport method to UDP, TCP, or TLS.
You cannot enter a string for this property.
Default: to-uri Values: request-uri | to-uri | from-uri | omit | UDP | TCP | TLS | string
Example: set transport omit
use-param: Specifies whether the User parameter in the TO URI of the SIP header is maintained or removed when the system forwards a message. If set to keep, the message is forwarded with the parameter as it was received. If set to omit, the entire user=param is removed from the TO URI.
Default: omit
Values: omit | keep
Example: set use-param keep
user-truncate-non-digits: Specifies whether to remove non-digits from the User portion of the TO URI in INVITE messages. When enabled, the system removes all non-digits.
Default: disabled
Values: enabled | disabled
Example: set user-truncate-non-digits enabled
strip-digits: Specifies the number of digits to strip from the User portion of the To URI. Use this, for example, if the original INVITE contains extra digits that would be problematic to the downstream server. Digits are removed beginning at the string that immediately follows the sip: or sips: portion.
Default: 0
Example: set strip-digits 2
The transcoding policy object allows you to configure the transcoding policy for the ME. This includes defining the preferred codec as deduced from SDP offers and answers, adapting to match received codecs, and rewriting rfc-2833 headers when encoding audio.
media-types: Sets a threshold, in megabytes, at which the system no longer writes recorded calls or IM files to the disk drive. The system sends a warning message to the event log (and an SNMP trap) indicating that space on the internal disk drive has been exceeded. The system checks the fail threshold each time it receives a call.
Default: There is no default setting
Values: pcma | pcmu | g7221 | g723 | g728 | g729 | g726-16 | g726-24 | g726-30 | g726-32 | gsm | gsm-ami | iLBC
Example: set media-types gsm
most-preferred: When true, the ME forces audio to only use the most preferred codec.
Default: false
Values: true | false
Example: set most-preferred true
symmetric-codec: When true, the ME adapts and matches the correct codec when the endpoint has switched the ”primary” codec.
Default: false
Values: true | false
Example: set symmetric-codec true
balance-ptime: When true, the ME attempts to balance RTP packet times with the SDP.
Default: true
Values: true | false
Example: set balance-ptime false
auto-release: When true, the ME attempts to release transcode resources when auto-anchoring is enabled.
Default: true
Values: true | false
Example: set auto-release false
block-unknown: When true, the ME blocks unnegotiated packet types.
Default: false
Values: true | false
Example: set block-unknown true
decode-telephone-events: When true, the ME decodes telephone-events into audio during transcoding when both sides do not support telephone-events.
Default: false
Values: true | false
Example: set decode-telephone-events true
Sets an interface to allow non-LCS devices to interact with LCS clients. The interface is a reference to a previously configured enterprise server: the LCS server that the ME uses as a gateway for non-LCS traffic. The server should be configured to treat the ME as authenticated.
For example, this interface would allow a SNOM phone to call an LCS Windows Messenger client. When an INVITE comes in from the SNOM phone, if the policy has the trusted-server property configured, the ME forwards the INVITE to that server. Since the server has been configured to ”treat as authenticated” traffic received from the ME on that interface, it does not prompt the SNOM phone, for authentication. The INVITE is forwarded to the Windows Messenger client, and the client sees the incoming call.
config vsp default-session-config trusted-interface-settings config vsp policies session-policies policy name rule name session-config trusted-interface-settings config vsp dial-plan dial-prefix entryName session-config trusted-interface-settings config vsp dial-plan route name session-config trusted-interface-settings config vsp dial-plan source-route name session-config trusted-interface-settings config vsp session-config-pool entry name trusted-interface-settings
Makes modifications to the user-to-user information (UUI) header. The UUI can be used for passing the universal call ID (UCID) and other session information to the NICE media server. If this object is configured, when the ME receives a SIP message it checks for the UUI header and applies the action defined with the replace-existing-header property. If no UUI header exists, the ME creates one.
config vsp default-session-config uui-header config vsp policies session-policies policy name rule name session-config uui-header config vsp dial-plan dial-prefix entryName session-config uui-header config vsp dial-plan route name session-config uui-header config vsp dial-plan source-route name session-config uui-header config vsp session-config-pool entry name uui-header
admin: Enables or disables this header manipulation configuration entry.
Default: disabled
Values: enabled | disabled
Example: set admin enabled
node-id: Specifies the number to be written to the UUI header, identifying the system to the media server. This is an internal integer specific to the ME.
Default: 0
Values: Min: 1 / Max: 65535
Example: set node-id 123
replace-existing-header: Specifies the action the system should take if there is an existing UUI header. If disabled, the system appends the modified UUI header to the existing one. If enabled, the system replaces the existing header with the new, modified version.
Default: disabled
Values: enabled | disabled
Example: set replace-existing-header enabled
You can move from an existing PBX infrastructure onto the ME, while still preserving the legacy PBX dial-plan and extension number configuration.
By configuring a Virtual Dial Plan (VDP), you can virtualize the PBX onto the ME. Users that are part of the same VDP are able to reach each other via extension dialing. Any dial plan routes and digit manipulations that existed on the legacy system are applied on the ME as they would have been on the PBX.
Via the vsp > virtual-dial-plan-pool object, configure virtual-dial-plans along with the dial-prefix, normalization, source-normalization, arbiter, route, and source-route elements for each. These objects are configured the same as vsp > dial-plan objects. For more information on configuring VDPs, see the Dial Plan Objects.
To determine a user's VDP assignment, configure the virtual-dialplan-settings. Specify the assigned VDP in the entry property. You can also specify a match-limit in case you configure the entry points and dial plans incorrectly and a VDP loop occurs.
vsp session-config-pool entry virtual-dialplan-settings vsp default-session-config virtual-dialplan-settings
entry: Specify the assigned VDP for this session-config.
Default: There is no default setting
Example: set entry vsp\tls\certificate\vdp1
match-limit: You can also specify a match-limit in case you configure the entry points and dial plans incorrectly and a virtual dial plan loop occurs.
Default: 100
Values: Min: 0 / Max: 65536
Example: set match-limit 150